
From nobody Thu Oct  2 06:36:49 2014
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From: Brett Tate <brett@broadsoft.com>
References: <78CFBBA55ECDC0ietf.shinji@gmail.com> <53F39858.9000106@alum.mit.edu> <FFCFBC226B9DC1ietf.shinji@gmail.com>
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Date: Thu, 2 Oct 2014 09:36:41 -0400
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To: OKUMURA Shinji <ietf.shinji@gmail.com>, sipcore@ietf.org
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Subject: Re: [sipcore] I-D Action: draft-sparks-sipcore-refer-explicit-subscription-00.txt
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> I just come up with another approach.
>
> A referrer send REFER within an existing dialog.
> And a notifier(referee) send NOTIFY within new dialog.
> Of course Request URI of NOTIFY is a GRUU of a referrer.
>
> It seems to be a simple solution for an avoidance of dialog-reuse.
>
> I think this option(Required: subscription-newdialog?) is available
> for not only REFER but also SUBSCRIBE.

One of the obstacles to this approach is that it would likely require an
update to RFC 6665.  The NOTIFY is not really a dialog-creating method.
RFC 6665 also significantly restricted what could create a subscription;
I'm not sure that this approach would satisfy the RFC 6665 restriction.

-brett


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From: Brett Tate <brett@broadsoft.com>
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Subject: [sipcore] draft-sparks-sipcore-refer-clarifications-04: comment
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Hi,

Section 3 indicates the following:

"3.  Use of GRUU is mandatory

   Section 4.5.1 of [RFC6665] makes GRUU [RFC5627] mandatory to
   implement and use as the local target in the subscription created by
   the REFER request."

It is hard not to recommend adding the following text to the section. ;)

"The mandate ensures that GRUU and the related patents are applicable to
as many devices as possible.  At least one Patent Holder states that its
position with respect to licensing any patent claims contained in the
disclosed patent would necessarily be infringed by implementation of the
technology required by the relevant IETF specification, for the purpose of
implementing such specification, is as follows:  Reasonable and
Non-Discriminatory License to All Implementers with Possible Royalty/Fee."

https://datatracker.ietf.org/ipr/1226/

https://datatracker.ietf.org/ipr/1832/

Cheers,
Brett


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From: OKUMURA Shinji <ietf.shinji@gmail.com>
To: Robert Sparks <rjsparks@nostrum.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Fri, 03 Oct 2014 17:40:53 +0900
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Subject: Re: [sipcore] Fwd: New Version Notification for draft-sparks-sipcore-refer-explicit-subscription-01.txt
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Hi,

As a general rule, REFER request may fork.
But a referrer receives only one response(i.e. Refer-Events-At).
Does this draft disallow a REFER request to be forked?

Regards,
Shinji

>-------- Forwarded Message -------- Subject: New Version Notification for 
>draft-sparks-sipcore-refer-explicit-subscription-01.txt
> Date: Mon, 08 Sep 2014 13:31:32 -0700
> From: [a:mailto:internet-drafts@ietf.org]internet-drafts@ietf.org
> To: Robert Sparks [a:mailto:rjsparks@nostrum.com]<rjsparks@nostrum.com>, 
>Robert Sparks [a:mailto:rjsparks@nostrum.com]<rjsparks@nostrum.com>
>
>
>A new version of I-D, draft-sparks-sipcore-refer-explicit-subscription-01.txt
>has been successfully submitted by Robert Sparks and posted to the
>IETF repository.
>
>Name:		draft-sparks-sipcore-refer-explicit-subscription
>Revision:	01
>Title:		Explicit Subscriptions for the REFER Method
>Document date:	2014-09-08
>Group:		Individual Submission
>Pages:		11
>URL:            [a:http://www.ietf.org/internet-drafts/draft-sparks-sipcore-
>refer-explicit-subscription-01.txt]http://www.ietf.org/internet-drafts/draft-
>sparks-sipcore-refer-explicit-subscription-01.txt
>Status:         [a:https://datatracker.ietf.org/doc/draft-sparks-sipcore-
>refer-explicit-subscription/]https://datatracker.ietf.org/doc/draft-sparks-
>sipcore-refer-explicit-subscription/
>Htmlized:       [a:http://tools.ietf.org/html/draft-sparks-sipcore-refer-
>explicit-subscription-01]http://tools.ietf.org/html/draft-sparks-sipcore-
>refer-explicit-subscription-01
>Diff:           [a:http://www.ietf.org/rfcdiff?url2=draft-sparks-sipcore-
>refer-explicit-subscription-01]http://www.ietf.org/rfcdiff?url2=draft-sparks-
>sipcore-refer-explicit-subscription-01
>
>Abstract:
>   The SIP REFER request, as defined by RFC3515, triggers an implicit
>   SIP-Specific Event Notification framework subscription.  Conflating
>   the start of the subscription with handling the REFER request makes
>   negotiating SUBSCRIBE extensions impossible, and complicates avoiding
>   SIP dialog sharing.  This document defines an extension to REFER to
>   remove the implicit subscription and replace it with an explicit one.
>
>                                                                               
>   
>
>
>Please note that it may take a couple of minutes from the time of submission
>until the htmlized version and diff are available at tools.ietf.org.
>
>The IETF Secretariat


From nobody Fri Oct  3 01:47:58 2014
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On 10/3/14 10:40 AM, OKUMURA Shinji wrote:
> Hi,
>
> As a general rule, REFER request may fork.
> But a referrer receives only one response(i.e. Refer-Events-At).
> Does this draft disallow a REFER request to be forked?
No.

If the REFER forks, it will get responses from each branch of the fork.
The response from each branch will have its own information about whether
the REFER was accepted on that branch, including its own Refer-Events-At
URI.

With the typical use of REFER specified to date (transfer), the refer 
will have
a Target-Dialog header field and will likely be accepted at only one of the
places it might have forked to.

RjS
>
> Regards,
> Shinji
>
>> -------- Forwarded Message -------- Subject: New Version Notification for
>> draft-sparks-sipcore-refer-explicit-subscription-01.txt
>> Date: Mon, 08 Sep 2014 13:31:32 -0700
>> From: [a:mailto:internet-drafts@ietf.org]internet-drafts@ietf.org
>> To: Robert Sparks [a:mailto:rjsparks@nostrum.com]<rjsparks@nostrum.com>,
>> Robert Sparks [a:mailto:rjsparks@nostrum.com]<rjsparks@nostrum.com>
>>
>>
>> A new version of I-D, draft-sparks-sipcore-refer-explicit-subscription-01.txt
>> has been successfully submitted by Robert Sparks and posted to the
>> IETF repository.
>>
>> Name:		draft-sparks-sipcore-refer-explicit-subscription
>> Revision:	01
>> Title:		Explicit Subscriptions for the REFER Method
>> Document date:	2014-09-08
>> Group:		Individual Submission
>> Pages:		11
>> URL:            [a:http://www.ietf.org/internet-drafts/draft-sparks-sipcore-
>> refer-explicit-subscription-01.txt]http://www.ietf.org/internet-drafts/draft-
>> sparks-sipcore-refer-explicit-subscription-01.txt
>> Status:         [a:https://datatracker.ietf.org/doc/draft-sparks-sipcore-
>> refer-explicit-subscription/]https://datatracker.ietf.org/doc/draft-sparks-
>> sipcore-refer-explicit-subscription/
>> Htmlized:       [a:http://tools.ietf.org/html/draft-sparks-sipcore-refer-
>> explicit-subscription-01]http://tools.ietf.org/html/draft-sparks-sipcore-
>> refer-explicit-subscription-01
>> Diff:           [a:http://www.ietf.org/rfcdiff?url2=draft-sparks-sipcore-
>> refer-explicit-subscription-01]http://www.ietf.org/rfcdiff?url2=draft-sparks-
>> sipcore-refer-explicit-subscription-01
>>
>> Abstract:
>>    The SIP REFER request, as defined by RFC3515, triggers an implicit
>>    SIP-Specific Event Notification framework subscription.  Conflating
>>    the start of the subscription with handling the REFER request makes
>>    negotiating SUBSCRIBE extensions impossible, and complicates avoiding
>>    SIP dialog sharing.  This document defines an extension to REFER to
>>    remove the implicit subscription and replace it with an explicit one.
>>
>>                                                                                
>>    
>>
>>
>> Please note that it may take a couple of minutes from the time of submission
>> until the htmlized version and diff are available at tools.ietf.org.
>>
>> The IETF Secretariat


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From: OKUMURA Shinji <ietf.shinji@gmail.com>
To: Brett Tate <brett@broadsoft.com>, sipcore@ietf.org
Date: Fri, 03 Oct 2014 18:35:41 +0900
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Subject: Re: [sipcore] I-D Action: draft-sparks-sipcore-refer-explicit-subscription-00.txt
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Hi, Brett.

Thanks for your reply.

>> I just come up with another approach.
>>
>> A referrer send REFER within an existing dialog.
>> And a notifier(referee) send NOTIFY within new dialog.
>> Of course Request URI of NOTIFY is a GRUU of a referrer.
>>
>> It seems to be a simple solution for an avoidance of dialog-reuse.
>>
>> I think this option(Required: subscription-newdialog?) is available
>> for not only REFER but also SUBSCRIBE.
>
>One of the obstacles to this approach is that it would likely require an
>update to RFC 6665.  The NOTIFY is not really a dialog-creating method.

RFC 6665 seems to say that a NOTIFY create a dialog, but
a SUBSCRIBE does not create a dialog.

Is my understanding wrong?

>RFC 6665 also significantly restricted what could create a subscription;
>I'm not sure that this approach would satisfy the RFC 6665 restriction.

Yes, my approach could be contrary to something else.
(i.e. section 4.4.1)

Regards,
Shinji


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We had a great event at SIPit 31. The summary is available at 
<https://www.sipit.net/SIPit31_summary>

RjS


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> >> A referrer send REFER within an existing dialog.
> >> And a notifier(referee) send NOTIFY within new dialog.
> >> Of course Request URI of NOTIFY is a GRUU of a referrer.
> >>
> >> It seems to be a simple solution for an avoidance of dialog-reuse.
> >>
> >> I think this option(Required: subscription-newdialog?) is available
> >> for not only REFER but also SUBSCRIBE.
> >
> >One of the obstacles to this approach is that it would likely require
> >an update to RFC 6665.  The NOTIFY is not really a dialog-creating
method.
>
> RFC 6665 seems to say that a NOTIFY create a dialog, but a SUBSCRIBE
does
> not create a dialog.
>
> Is my understanding wrong?

Section 4.1.2.1 indicates "SUBSCRIBE is a dialog-creating method, as
described in [RFC3261]".

The NOTIFY contains a To tag which is one of the items that allow it be
matched to the SUBSCRIBE or REFER.

Section 4.4.1:

"NOTIFY requests are matched to such SUBSCRIBE requests if they
  contain the same "Call-ID", a "To" header field "tag" parameter that
  matches the "From" header field "tag" parameter of the SUBSCRIBE
  request, and the same "Event" header field."


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From: OKUMURA Shinji <ietf.shinji@gmail.com>
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Subject: Re: [sipcore] New Version Notification for draft-sparks-sipcore-refer-explicit-subscription-01
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Hello Robert,

>On 10/3/14 10:40 AM, OKUMURA Shinji wrote:
>> Hi,
>>
>> As a general rule, REFER request may fork.
>> But a referrer receives only one response(i.e. Refer-Events-At).
>> Does this draft disallow a REFER request to be forked?
>No.
>
>If the REFER forks, it will get responses from each branch of the fork.
>The response from each branch will have its own information about whether
>the REFER was accepted on that branch, including its own Refer-Events-At
>URI.

In accordance with the following, a successful response for REFER
request is only one.

RFC3261
17.1.2.1 Overview of the non-INVITE Transaction

   Unlike an INVITE transaction, a non-INVITE transaction has no special
   handling for the 2xx response.  The result is that only a single 2xx
   response to a non-INVITE is ever delivered to a UAC.


>With the typical use of REFER specified to date (transfer), the refer 
>will have
>a Target-Dialog header field and will likely be accepted at only one of the
>places it might have forked to.
>
>RjS

Regards,
Shinji


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From: OKUMURA Shinji <ietf.shinji@gmail.com>
To: Brett Tate <brett@broadsoft.com>, sipcore@ietf.org
Date: Mon, 06 Oct 2014 15:24:55 +0900
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Subject: Re: [sipcore] I-D Action: draft-sparks-sipcore-refer-explicit-subscription-00.txt
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Hi Brett,

>> >> A referrer send REFER within an existing dialog.
>> >> And a notifier(referee) send NOTIFY within new dialog.
>> >> Of course Request URI of NOTIFY is a GRUU of a referrer.
>> >>
>> >> It seems to be a simple solution for an avoidance of dialog-reuse.
>> >>
>> >> I think this option(Required: subscription-newdialog?) is available
>> >> for not only REFER but also SUBSCRIBE.
>> >
>> >One of the obstacles to this approach is that it would likely require
>> >an update to RFC 6665.  The NOTIFY is not really a dialog-creating method.
>>
>> RFC 6665 seems to say that a NOTIFY create a dialog, but a SUBSCRIBE does
>> not create a dialog.
>>
>> Is my understanding wrong?
>
>Section 4.1.2.1 indicates "SUBSCRIBE is a dialog-creating method, as
>described in [RFC3261]".
>
>The NOTIFY contains a To tag which is one of the items that allow it be
>matched to the SUBSCRIBE or REFER.
>
>Section 4.4.1:
>
>"NOTIFY requests are matched to such SUBSCRIBE requests if they
>  contain the same "Call-ID", a "To" header field "tag" parameter that
>  matches the "From" header field "tag" parameter of the SUBSCRIBE
>  request, and the same "Event" header field."

But then,
Appendix B.  Changes from RFC 3265
B.22.  Rationalize Dialog Creation
   Section 4.4.1 has been updated to specify that dialogs should be
   created when the NOTIFY arrives.  Previously, the dialog was
   established by the SUBSCRIBE 200 or by the NOTIFY transaction.  This
   was unnecessarily complicated; the newer rules are easier to
   implement (and result in effectively the same behavior on the wire).

For me, these descriptions are considerably ambiguous.

Regards,
Shinji


From nobody Mon Oct  6 05:41:42 2014
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--089e0141ab444d6df80504c06712
Content-Type: text/plain; charset=ISO-8859-1

Hi,

As part of the SIP Over IPv6 Task Group activities in the SIP Forum we
created the following *Informational* draft which describes the impact on
IPv4 SIP elements when interacting with IPv6 SIP elements.
The draft was initially discussed on the DISPATCH mailing list, but after
discussing this draft with Mary Barnes (co-chair of DISPATCH) and Paul
Kyzivat (co-chair of SIPCore) it was decided that the SIPCore would the
right place to discuss this.

We would appreciate it if people review and provide feedback on this draft.

Regards,
 Rifaat




-----Original Message-----
From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
Sent: Sunday, October 05, 2014 7:44 PM
To: Shekh-Yusef, Rifaat (Rifaat); Carl Klatsky; Gonzalo Salgueiro; Andrew
Hutton; Olle E.Johansson; Olle E. Johansson; Carl Klatsky; Gonzalo
Salgueiro; Shekh-Yusef, Rifaat (Rifaat); Andrew Hutton
Subject: New Version Notification for
draft-klatsky-sipcore-ipv6-impact-ipv4-00.txt


A new version of I-D, draft-klatsky-sipcore-ipv6-impact-ipv4-00.txt
has been successfully submitted by Carl Klatsky and posted to the IETF
repository.

Name:           draft-klatsky-sipcore-ipv6-impact-ipv4
Revision:       00
Title:          Interoperability Impacts of IPv6 Interworking with Existing
IPv4 SIP Implementations
Document date:  2014-10-05
Group:          Individual Submission
Pages:          10
URL:
http://www.ietf.org/internet-drafts/draft-klatsky-sipcore-ipv6-impact-ipv4-00.txt
Status:
https://datatracker.ietf.org/doc/draft-klatsky-sipcore-ipv6-impact-ipv4/
Htmlized:
http://tools.ietf.org/html/draft-klatsky-sipcore-ipv6-impact-ipv4-00


Abstract:
   This document captures potential impacts to IPv4 SIP implementations
   when interworking with IPv6 SIP implementations.  Although some
   amount of interworking translation will occur at the network and
   application layers, an IPv4 SIP application may still encounter a SIP
   message with some IPv6 values in it, resulting in unforeseen error
   conditions.  Such potential scenarios will be identified in this
   document so that SIP application developers can define solutions to
   handle these cases.  Note, this document is not intended to be an
   exhaustive list, rather to provide an overview of some of the more
   commonly encountered potential scenarios.




Please note that it may take a couple of minutes from the time of
submission until the htmlized version and diff are available at
tools.ietf.org.

The IETF Secretariat

--089e0141ab444d6df80504c06712
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<div dir=3D"ltr"><div class=3D"gmail_quote">Hi,</div><div class=3D"gmail_qu=
ote"><br></div><div class=3D"gmail_quote">As part of the SIP Over IPv6 Task=
 Group activities in the SIP Forum we created the following <b>Informationa=
l</b> draft which describes the impact on IPv4 SIP elements when interactin=
g with IPv6 SIP elements.</div><div class=3D"gmail_quote">The draft was ini=
tially discussed on the DISPATCH mailing list, but after discussing this dr=
aft with Mary Barnes (co-chair of DISPATCH) and Paul Kyzivat (co-chair of S=
IPCore) it was decided that the SIPCore would the right place to discuss th=
is.</div><div class=3D"gmail_quote"><br></div><div class=3D"gmail_quote">We=
 would appreciate it if people review and provide feedback on this draft.</=
div><div class=3D"gmail_quote"><br></div><div class=3D"gmail_quote">Regards=
,</div><div class=3D"gmail_quote">=A0Rifaat</div><div class=3D"gmail_quote"=
><br></div><div class=3D"gmail_quote"><br><br>
<br>
-----Original Message-----<br>
From: <a href=3D"mailto:internet-drafts@ietf.org" target=3D"_blank">interne=
t-drafts@ietf.org</a> [mailto:<a href=3D"mailto:internet-drafts@ietf.org" t=
arget=3D"_blank">internet-drafts@ietf.org</a>]<br>
Sent: Sunday, October 05, 2014 7:44 PM<br>
To: Shekh-Yusef, Rifaat (Rifaat); Carl Klatsky; Gonzalo Salgueiro; Andrew H=
utton; Olle E.Johansson; Olle E. Johansson; Carl Klatsky; Gonzalo Salgueiro=
; Shekh-Yusef, Rifaat (Rifaat); Andrew Hutton<br>
Subject: New Version Notification for draft-klatsky-sipcore-ipv6-impact-ipv=
4-00.txt<br>
<br>
<br>
A new version of I-D, draft-klatsky-sipcore-ipv6-impact-ipv4-00.txt<br>
has been successfully submitted by Carl Klatsky and posted to the IETF repo=
sitory.<br>
<br>
Name:=A0 =A0 =A0 =A0 =A0 =A0draft-klatsky-sipcore-ipv6-impact-ipv4<br>
Revision:=A0 =A0 =A0 =A000<br>
Title:=A0 =A0 =A0 =A0 =A0 Interoperability Impacts of IPv6 Interworking wit=
h Existing IPv4 SIP Implementations<br>
Document date:=A0 2014-10-05<br>
Group:=A0 =A0 =A0 =A0 =A0 Individual Submission<br>
Pages:=A0 =A0 =A0 =A0 =A0 10<br>
URL:=A0 =A0 =A0 =A0 =A0 =A0 <a href=3D"http://www.ietf.org/internet-drafts/=
draft-klatsky-sipcore-ipv6-impact-ipv4-00.txt" target=3D"_blank">http://www=
.ietf.org/internet-drafts/draft-klatsky-sipcore-ipv6-impact-ipv4-00.txt</a>=
<br>
Status:=A0 =A0 =A0 =A0 =A0<a href=3D"https://datatracker.ietf.org/doc/draft=
-klatsky-sipcore-ipv6-impact-ipv4/" target=3D"_blank">https://datatracker.i=
etf.org/doc/draft-klatsky-sipcore-ipv6-impact-ipv4/</a><br>
Htmlized:=A0 =A0 =A0 =A0<a href=3D"http://tools.ietf.org/html/draft-klatsky=
-sipcore-ipv6-impact-ipv4-00" target=3D"_blank">http://tools.ietf.org/html/=
draft-klatsky-sipcore-ipv6-impact-ipv4-00</a><br>
<br>
<br>
Abstract:<br>
=A0 =A0This document captures potential impacts to IPv4 SIP implementations=
<br>
=A0 =A0when interworking with IPv6 SIP implementations.=A0 Although some<br=
>
=A0 =A0amount of interworking translation will occur at the network and<br>
=A0 =A0application layers, an IPv4 SIP application may still encounter a SI=
P<br>
=A0 =A0message with some IPv6 values in it, resulting in unforeseen error<b=
r>
=A0 =A0conditions.=A0 Such potential scenarios will be identified in this<b=
r>
=A0 =A0document so that SIP application developers can define solutions to<=
br>
=A0 =A0handle these cases.=A0 Note, this document is not intended to be an<=
br>
=A0 =A0exhaustive list, rather to provide an overview of some of the more<b=
r>
=A0 =A0commonly encountered potential scenarios.<br>
<br>
<br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n until the htmlized version and diff are available at <a href=3D"http://to=
ols.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
The IETF Secretariat<br>
<br>
</div><br></div>

--089e0141ab444d6df80504c06712--


From nobody Mon Oct  6 06:43:20 2014
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Hi!
Just updated the version number to make this draft active again.

During SIPit31 last week we confirmed the issues mentioned in the draft, =
but also discovered a new issue.=20

This may not be directly SIP related but important for developers:

We added a few IPv6 ULA networks to the SIPit LAN. It seemed like =
multiple network addresses apart from one global and one link-local had =
not been tested by the developers before. We noticed that =
implementations had problems selecting candidates for ICE as well as =
selecting source IP address. I added a SIP proxy on a ULA address and it =
could not be reached. Aside from that, the proxy itself had issues =
selecting source address (with a more than ten year old IPv6 =
implementation). It made it very clear to participants that "one =
interface, one address" does no longer apply.

Cheers,
/O



Begin forwarded message:

> From: internet-drafts@ietf.org
> Subject: New Version Notification for =
draft-johansson-sip-dual-stack-03.txt
> Date: 6 Oct 2014 15:27:14 GMT+2
> To: "Gonzalo Salgueiro" <gsalguei@cisco.com>, Olle E. Johansson =
<oej@edvina.net>, "Olle E. Johansson" <oej@edvina.net>, Gonzalo =
Salgueiro <gsalguei@cisco.com>
>=20
>=20
> A new version of I-D, draft-johansson-sip-dual-stack-03.txt
> has been successfully submitted by Olle E. Johansson and posted to the
> IETF repository.
>=20
> Name:		draft-johansson-sip-dual-stack
> Revision:	03
> Title:		Locating Session Initiation Protocol (SIP) =
Servers in a Dual-Stack IP Network
> Document date:	2014-10-06
> Group:		Individual Submission
> Pages:		6
> URL:            =
http://www.ietf.org/internet-drafts/draft-johansson-sip-dual-stack-03.txt
> Status:         =
https://datatracker.ietf.org/doc/draft-johansson-sip-dual-stack/
> Htmlized:       =
http://tools.ietf.org/html/draft-johansson-sip-dual-stack-03
> Diff:           =
http://www.ietf.org/rfcdiff?url2=3Ddraft-johansson-sip-dual-stack-03
>=20
> Abstract:
>   RFC 3263 defines how a Session Initiation Protocol (SIP)
>   implementation, given a SIP Uniform Resource Identifier (URI), =
should
>   locate the next hop SIP server using Domain Name System (DNS)
>   procedures.  The specification repeatedly states that the
>   implementation should look up IPv4 or IPv6 addresses.  This is not a
>   suitable solution and one that can cause severely degraded user
>   experience dual-stack clients, as detailed in the Happy Eyeballs
>   specification.  This document specifies amended procedures for dual-
>   stack SIP implementations so that they look up both IPv4 and IPv6
>   addresses.  This way, the SIP implementation can find the preferred
>   network path and protocol with an improved chance of successfully
>   reaching the desired service.  This document also clarifies DNS SRV
>   usage for single-stack clients.
>=20
>=20
>=20
>=20
> Please note that it may take a couple of minutes from the time of =
submission
> until the htmlized version and diff are available at tools.ietf.org.
>=20
> The IETF Secretariat
>=20


From nobody Mon Oct  6 07:21:05 2014
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Subject: [sipcore] Fwd: New Version Notification for draft-johansson-sipcore-dane-sip-00.txt
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Just a update of the name - no content changed - to keep it on track.

Cheers,
/O

Begin forwarded message:

> From: internet-drafts@ietf.org
> Subject: New Version Notification for =
draft-johansson-sipcore-dane-sip-00.txt
> Date: 6 Oct 2014 16:16:17 GMT+2
> To: Olle E. Johansson <oej@edvina.net>, "Olle E. Johansson" =
<oej@edvina.net>
>=20
>=20
> A new version of I-D, draft-johansson-sipcore-dane-sip-00.txt
> has been successfully submitted by Olle E. Johansson and posted to the
> IETF repository.
>=20
> Name:		draft-johansson-sipcore-dane-sip
> Revision:	00
> Title:		TLS sessions in SIP using DNS-based =
Authentication of Named Entities (DANE) TLSA records
> Document date:	2014-10-06
> Group:		Individual Submission
> Pages:		9
> URL:            =
http://www.ietf.org/internet-drafts/draft-johansson-sipcore-dane-sip-00.tx=
t
> Status:         =
https://datatracker.ietf.org/doc/draft-johansson-sipcore-dane-sip/
> Htmlized:       =
http://tools.ietf.org/html/draft-johansson-sipcore-dane-sip-00
>=20
>=20
> Abstract:
>   Use of TLS in the SIP protocol is defined in multiple documents,
>   starting with RFC 3261.  The actual verification that happens when
>   setting up a SIP TLS connection to a SIP server based on a SIP URI =
is
>   described in detail in RFC 5922 - SIP Domain Certificates.
>=20
>   In this document, an alternative method is defined, using DNS-Based
>   Authentication of Named Entities (DANE).  By looking up TLSA DNS
>   records and using DNSsec protection of the required queries,
>   including lookups for NAPTR and SRV records, a SIP Client can verify
>   the identity of the TLS SIP server in a different way, matching on
>   the SRV host name in the X.509 PKIX certificate instead of the SIP
>   domain.  This provides more scalability in hosting solutions and =
make
>   it easier to use standard CA certificates (if needed at all).
>=20
>=20
>=20
>=20
> Please note that it may take a couple of minutes from the time of =
submission
> until the htmlized version and diff are available at tools.ietf.org.
>=20
> The IETF Secretariat
>=20


From nobody Mon Oct  6 07:26:11 2014
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Subject: Re: [sipcore] New Version Notification for draft-sparks-sipcore-refer-explicit-subscription-01
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On 10/6/14 12:44 AM, OKUMURA Shinji wrote:
> Hello Robert,
>
>> On 10/3/14 10:40 AM, OKUMURA Shinji wrote:
>>> Hi,
>>>
>>> As a general rule, REFER request may fork.
>>> But a referrer receives only one response(i.e. Refer-Events-At).
>>> Does this draft disallow a REFER request to be forked?
>> No.
>>
>> If the REFER forks, it will get responses from each branch of the fork.
>> The response from each branch will have its own information about whether
>> the REFER was accepted on that branch, including its own Refer-Events-At
>> URI.
> In accordance with the following, a successful response for REFER
> request is only one.
>
> RFC3261
> 17.1.2.1 Overview of the non-INVITE Transaction
>
>     Unlike an INVITE transaction, a non-INVITE transaction has no special
>     handling for the 2xx response.  The result is that only a single 2xx
>     response to a non-INVITE is ever delivered to a UAC.
Wow (slaps forehead) - very good catch. That's the whole reason the 
immediate NOTIFY
in SIP Events exists.

The consequences of that are going to require some time to think through.


>
>
>> With the typical use of REFER specified to date (transfer), the refer
>> will have
>> a Target-Dialog header field and will likely be accepted at only one of the
>> places it might have forked to.
>>
>> RjS
> Regards,
> Shinji


From nobody Mon Oct  6 07:33:29 2014
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Subject: Re: [sipcore] Fwd: New Version Notification for draft-johansson-sip-dual-stack-03.txt
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On 06 Oct 2014, at 15:43, Olle E. Johansson <oej@edvina.net> wrote:

> Hi!
> Just updated the version number to make this draft active again.
Which was a stupid thing, since it's replaced by =
draft-ietf-sipcore-dns-dual-stack-00.

Life.

/O
>=20
> During SIPit31 last week we confirmed the issues mentioned in the =
draft, but also discovered a new issue.=20
>=20
> This may not be directly SIP related but important for developers:
>=20
> We added a few IPv6 ULA networks to the SIPit LAN. It seemed like =
multiple network addresses apart from one global and one link-local had =
not been tested by the developers before. We noticed that =
implementations had problems selecting candidates for ICE as well as =
selecting source IP address. I added a SIP proxy on a ULA address and it =
could not be reached. Aside from that, the proxy itself had issues =
selecting source address (with a more than ten year old IPv6 =
implementation). It made it very clear to participants that "one =
interface, one address" does no longer apply.
>=20
> Cheers,
> /O
>=20
>=20
>=20
> Begin forwarded message:
>=20
>> From: internet-drafts@ietf.org
>> Subject: New Version Notification for =
draft-johansson-sip-dual-stack-03.txt
>> Date: 6 Oct 2014 15:27:14 GMT+2
>> To: "Gonzalo Salgueiro" <gsalguei@cisco.com>, Olle E. Johansson =
<oej@edvina.net>, "Olle E. Johansson" <oej@edvina.net>, Gonzalo =
Salgueiro <gsalguei@cisco.com>
>>=20
>>=20
>> A new version of I-D, draft-johansson-sip-dual-stack-03.txt
>> has been successfully submitted by Olle E. Johansson and posted to =
the
>> IETF repository.
>>=20
>> Name:		draft-johansson-sip-dual-stack
>> Revision:	03
>> Title:		Locating Session Initiation Protocol (SIP) =
Servers in a Dual-Stack IP Network
>> Document date:	2014-10-06
>> Group:		Individual Submission
>> Pages:		6
>> URL:            =
http://www.ietf.org/internet-drafts/draft-johansson-sip-dual-stack-03.txt
>> Status:         =
https://datatracker.ietf.org/doc/draft-johansson-sip-dual-stack/
>> Htmlized:       =
http://tools.ietf.org/html/draft-johansson-sip-dual-stack-03
>> Diff:           =
http://www.ietf.org/rfcdiff?url2=3Ddraft-johansson-sip-dual-stack-03
>>=20
>> Abstract:
>>  RFC 3263 defines how a Session Initiation Protocol (SIP)
>>  implementation, given a SIP Uniform Resource Identifier (URI), =
should
>>  locate the next hop SIP server using Domain Name System (DNS)
>>  procedures.  The specification repeatedly states that the
>>  implementation should look up IPv4 or IPv6 addresses.  This is not a
>>  suitable solution and one that can cause severely degraded user
>>  experience dual-stack clients, as detailed in the Happy Eyeballs
>>  specification.  This document specifies amended procedures for dual-
>>  stack SIP implementations so that they look up both IPv4 and IPv6
>>  addresses.  This way, the SIP implementation can find the preferred
>>  network path and protocol with an improved chance of successfully
>>  reaching the desired service.  This document also clarifies DNS SRV
>>  usage for single-stack clients.
>>=20
>>=20
>>=20
>>=20
>> Please note that it may take a couple of minutes from the time of =
submission
>> until the htmlized version and diff are available at tools.ietf.org.
>>=20
>> The IETF Secretariat
>>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From nobody Mon Oct  6 07:59:44 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Robert Sparks <rjsparks@nostrum.com>, OKUMURA Shinji <ietf.shinji@gmail.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: [sipcore] New Version Notification for draft-sparks-sipcore-refer-explicit-subscription-01
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Subject: Re: [sipcore] New Version Notification for draft-sparks-sipcore-refer-explicit-subscription-01
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Hi,

This of course does not apply to in-dialog REFERs, where forking does not o=
ccur.

Regards,

Christer


Sent from my Windows Phone
________________________________
From: Robert Sparks<mailto:rjsparks@nostrum.com>
Sent: =FD06/=FD10/=FD2014 17:26
To: OKUMURA Shinji<mailto:ietf.shinji@gmail.com>; sipcore@ietf.org<mailto:s=
ipcore@ietf.org>
Subject: Re: [sipcore] New Version Notification for draft-sparks-sipcore-re=
fer-explicit-subscription-01


On 10/6/14 12:44 AM, OKUMURA Shinji wrote:
> Hello Robert,
>
>> On 10/3/14 10:40 AM, OKUMURA Shinji wrote:
>>> Hi,
>>>
>>> As a general rule, REFER request may fork.
>>> But a referrer receives only one response(i.e. Refer-Events-At).
>>> Does this draft disallow a REFER request to be forked?
>> No.
>>
>> If the REFER forks, it will get responses from each branch of the fork.
>> The response from each branch will have its own information about whethe=
r
>> the REFER was accepted on that branch, including its own Refer-Events-At
>> URI.
> In accordance with the following, a successful response for REFER
> request is only one.
>
> RFC3261
> 17.1.2.1 Overview of the non-INVITE Transaction
>
>     Unlike an INVITE transaction, a non-INVITE transaction has no special
>     handling for the 2xx response.  The result is that only a single 2xx
>     response to a non-INVITE is ever delivered to a UAC.
Wow (slaps forehead) - very good catch. That's the whole reason the
immediate NOTIFY
in SIP Events exists.

The consequences of that are going to require some time to think through.


>
>
>> With the typical use of REFER specified to date (transfer), the refer
>> will have
>> a Target-Dialog header field and will likely be accepted at only one of =
the
>> places it might have forked to.
>>
>> RjS
> Regards,
> Shinji

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<div style=3D"font-family:Calibri,sans-serif; font-size:11pt">Hi,<br>
<br>
This of course does not apply to in-dialog REFERs, where forking does not o=
ccur.<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
<br>
Sent from my Windows Phone</div>
</div>
<div dir=3D"ltr">
<hr>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">From:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:rjsparks@nostrum.com">Robert Sparks</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Sent:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">=FD06=
/=FD10/=FD2014 17:26</span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">To:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:ietf.shinji@gmail.com">OKUMURA Shinji</a>;
<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Subject:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">Re: [=
sipcore] New Version Notification for draft-sparks-sipcore-refer-explicit-s=
ubscription-01</span><br>
<br>
</div>
</div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText"><br>
On 10/6/14 12:44 AM, OKUMURA Shinji wrote:<br>
&gt; Hello Robert,<br>
&gt;<br>
&gt;&gt; On 10/3/14 10:40 AM, OKUMURA Shinji wrote:<br>
&gt;&gt;&gt; Hi,<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; As a general rule, REFER request may fork.<br>
&gt;&gt;&gt; But a referrer receives only one response(i.e. Refer-Events-At=
).<br>
&gt;&gt;&gt; Does this draft disallow a REFER request to be forked?<br>
&gt;&gt; No.<br>
&gt;&gt;<br>
&gt;&gt; If the REFER forks, it will get responses from each branch of the =
fork.<br>
&gt;&gt; The response from each branch will have its own information about =
whether<br>
&gt;&gt; the REFER was accepted on that branch, including its own Refer-Eve=
nts-At<br>
&gt;&gt; URI.<br>
&gt; In accordance with the following, a successful response for REFER<br>
&gt; request is only one.<br>
&gt;<br>
&gt; RFC3261<br>
&gt; 17.1.2.1 Overview of the non-INVITE Transaction<br>
&gt;<br>
&gt;&nbsp;&nbsp;&nbsp;&nbsp; Unlike an INVITE transaction, a non-INVITE tra=
nsaction has no special<br>
&gt;&nbsp;&nbsp;&nbsp;&nbsp; handling for the 2xx response.&nbsp; The resul=
t is that only a single 2xx<br>
&gt;&nbsp;&nbsp;&nbsp;&nbsp; response to a non-INVITE is ever delivered to =
a UAC.<br>
Wow (slaps forehead) - very good catch. That's the whole reason the <br>
immediate NOTIFY<br>
in SIP Events exists.<br>
<br>
The consequences of that are going to require some time to think through.<b=
r>
<br>
<br>
&gt;<br>
&gt;<br>
&gt;&gt; With the typical use of REFER specified to date (transfer), the re=
fer<br>
&gt;&gt; will have<br>
&gt;&gt; a Target-Dialog header field and will likely be accepted at only o=
ne of the<br>
&gt;&gt; places it might have forked to.<br>
&gt;&gt;<br>
&gt;&gt; RjS<br>
&gt; Regards,<br>
&gt; Shinji<br>
<br>
_______________________________________________<br>
sipcore mailing list<br>
sipcore@ietf.org<br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore">https://www.ietf.=
org/mailman/listinfo/sipcore</a><br>
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--_000_7594FB04B1934943A5C02806D1A2204B1D46A059ESESSMB209erics_--


From nobody Mon Oct  6 10:49:59 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Robert Sparks <rjsparks@nostrum.com>, OKUMURA Shinji <ietf.shinji@gmail.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: [sipcore] New Version Notification for draft-sparks-sipcore-refer-explicit-subscription-01
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Date: Mon, 6 Oct 2014 17:49:52 +0000
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Subject: Re: [sipcore] New Version Notification for draft-sparks-sipcore-refer-explicit-subscription-01
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Hi,

>>>> As a general rule, REFER request may fork.
>>>> But a referrer receives only one response(i.e. Refer-Events-At).
>>>> Does this draft disallow a REFER request to be forked?
>>> No.
>>>
>>> If the REFER forks, it will get responses from each branch of the fork.
>>> The response from each branch will have its own information about=20
>>> whether the REFER was accepted on that branch, including its own=20
>>> Refer-Events-At URI.
>> In accordance with the following, a successful response for REFER=20
>> request is only one.
>>
>> RFC3261
>> 17.1.2.1 Overview of the non-INVITE Transaction
>>
>>     Unlike an INVITE transaction, a non-INVITE transaction has no specia=
l
>>     handling for the 2xx response.  The result is that only a single 2xx
>>     response to a non-INVITE is ever delivered to a UAC.
>>
>
> Wow (slaps forehead) - very good catch. That's the whole reason the=20
> immediate NOTIFY in SIP Events exists.
>
> The consequences of that are going to require some time to think through.

One option would to still send the immediate NOTIFY, but with the "Subscrip=
tion-State" header field value set to "terminated".

(But, as said earlier, for in-dialog REFERs such NOTIFY would not be needed=
, as there is no forking.)

Regards,

Christer


From nobody Mon Oct  6 12:00:19 2014
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Subject: Re: [sipcore] New Version Notification for draft-sparks-sipcore-refer-explicit-subscription-01
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On 10/6/14 10:25 AM, Robert Sparks wrote:
>
> On 10/6/14 12:44 AM, OKUMURA Shinji wrote:
>> Hello Robert,
>>
>>> On 10/3/14 10:40 AM, OKUMURA Shinji wrote:
>>>> Hi,
>>>>
>>>> As a general rule, REFER request may fork.
>>>> But a referrer receives only one response(i.e. Refer-Events-At).
>>>> Does this draft disallow a REFER request to be forked?
>>> No.
>>>
>>> If the REFER forks, it will get responses from each branch of the fork.
>>> The response from each branch will have its own information about
>>> whether
>>> the REFER was accepted on that branch, including its own Refer-Events-At
>>> URI.
>> In accordance with the following, a successful response for REFER
>> request is only one.
>>
>> RFC3261
>> 17.1.2.1 Overview of the non-INVITE Transaction
>>
>>     Unlike an INVITE transaction, a non-INVITE transaction has no special
>>     handling for the 2xx response.  The result is that only a single 2xx
>>     response to a non-INVITE is ever delivered to a UAC.
> Wow (slaps forehead) - very good catch. That's the whole reason the
> immediate NOTIFY
> in SIP Events exists.
>
> The consequences of that are going to require some time to think through.

One way to handle this is to:

- allow multiple Refer-Events-At header fields in the response
- make it the responsibility of the forking proxy to gather the 
responses from each fork and consolidate them into a single response.

This isn't ideal. It requires the proxy to support the feature. And it 
means it must special case forking of REFER, since normally it can 
simply forward the first 2xx response it gets to a forked request.

OTOH, I'll guess there is very little current use of out-of-dialog 
REFER. So there may not be many backward compatibility issues with that.

	Thanks,
	Paul


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To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "sipcore@ietf.org" <sipcore@ietf.org>
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Archived-At: http://mailarchive.ietf.org/arch/msg/sipcore/9i9DXfGI4MdJgZET7O_prVs-Iwc
Subject: Re: [sipcore] New Version Notification for draft-sparks-sipcore-refer-explicit-subscription-01
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Hi,

I don't think we shall assume support in forking proxy (why didn't we defin=
e a Forking-Proxy-Require header field back in the days...).

I suggested to use
NOTIFY which indicates terminated. If that doesn't work, perhaps we should =
disallow the mechanism in out-of-dialog REFERs. After all, the need for the=
 mechanism comes from in-dialog usage of REFER...

Regards,

Christer

Sent from my Windows Phone
________________________________
From: Paul Kyzivat<mailto:pkyzivat@alum.mit.edu>
Sent: =FD06/=FD10/=FD2014 22:00
To: sipcore@ietf.org<mailto:sipcore@ietf.org>
Subject: Re: [sipcore] New Version Notification for draft-sparks-sipcore-re=
fer-explicit-subscription-01

On 10/6/14 10:25 AM, Robert Sparks wrote:
>
> On 10/6/14 12:44 AM, OKUMURA Shinji wrote:
>> Hello Robert,
>>
>>> On 10/3/14 10:40 AM, OKUMURA Shinji wrote:
>>>> Hi,
>>>>
>>>> As a general rule, REFER request may fork.
>>>> But a referrer receives only one response(i.e. Refer-Events-At).
>>>> Does this draft disallow a REFER request to be forked?
>>> No.
>>>
>>> If the REFER forks, it will get responses from each branch of the fork.
>>> The response from each branch will have its own information about
>>> whether
>>> the REFER was accepted on that branch, including its own Refer-Events-A=
t
>>> URI.
>> In accordance with the following, a successful response for REFER
>> request is only one.
>>
>> RFC3261
>> 17.1.2.1 Overview of the non-INVITE Transaction
>>
>>     Unlike an INVITE transaction, a non-INVITE transaction has no specia=
l
>>     handling for the 2xx response.  The result is that only a single 2xx
>>     response to a non-INVITE is ever delivered to a UAC.
> Wow (slaps forehead) - very good catch. That's the whole reason the
> immediate NOTIFY
> in SIP Events exists.
>
> The consequences of that are going to require some time to think through.

One way to handle this is to:

- allow multiple Refer-Events-At header fields in the response
- make it the responsibility of the forking proxy to gather the
responses from each fork and consolidate them into a single response.

This isn't ideal. It requires the proxy to support the feature. And it
means it must special case forking of REFER, since normally it can
simply forward the first 2xx response it gets to a forked request.

OTOH, I'll guess there is very little current use of out-of-dialog
REFER. So there may not be many backward compatibility issues with that.

        Thanks,
        Paul

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

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<div>
<div>
<div style=3D"font-family:Calibri,sans-serif; font-size:11pt">Hi,<br>
<br>
I don't think we shall assume support in forking proxy (why didn't we defin=
e a Forking-Proxy-Require header field back in the days...).
<br>
<br>
I suggested to use<br>
NOTIFY which indicates terminated. If that doesn't work, perhaps we should =
disallow the mechanism in out-of-dialog REFERs. After all, the need for the=
 mechanism comes from in-dialog usage of REFER...<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
Sent from my Windows Phone</div>
</div>
<div dir=3D"ltr">
<hr>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">From:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:pkyzivat@alum.mit.edu">Paul Kyzivat</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Sent:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">=FD06=
/=FD10/=FD2014 22:00</span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">To:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Subject:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">Re: [=
sipcore] New Version Notification for draft-sparks-sipcore-refer-explicit-s=
ubscription-01</span><br>
<br>
</div>
</div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">On 10/6/14 10:25 AM, Robert Sparks wrote:<br>
&gt;<br>
&gt; On 10/6/14 12:44 AM, OKUMURA Shinji wrote:<br>
&gt;&gt; Hello Robert,<br>
&gt;&gt;<br>
&gt;&gt;&gt; On 10/3/14 10:40 AM, OKUMURA Shinji wrote:<br>
&gt;&gt;&gt;&gt; Hi,<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; As a general rule, REFER request may fork.<br>
&gt;&gt;&gt;&gt; But a referrer receives only one response(i.e. Refer-Event=
s-At).<br>
&gt;&gt;&gt;&gt; Does this draft disallow a REFER request to be forked?<br>
&gt;&gt;&gt; No.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; If the REFER forks, it will get responses from each branch of =
the fork.<br>
&gt;&gt;&gt; The response from each branch will have its own information ab=
out<br>
&gt;&gt;&gt; whether<br>
&gt;&gt;&gt; the REFER was accepted on that branch, including its own Refer=
-Events-At<br>
&gt;&gt;&gt; URI.<br>
&gt;&gt; In accordance with the following, a successful response for REFER<=
br>
&gt;&gt; request is only one.<br>
&gt;&gt;<br>
&gt;&gt; RFC3261<br>
&gt;&gt; 17.1.2.1 Overview of the non-INVITE Transaction<br>
&gt;&gt;<br>
&gt;&gt;&nbsp;&nbsp;&nbsp;&nbsp; Unlike an INVITE transaction, a non-INVITE=
 transaction has no special<br>
&gt;&gt;&nbsp;&nbsp;&nbsp;&nbsp; handling for the 2xx response.&nbsp; The r=
esult is that only a single 2xx<br>
&gt;&gt;&nbsp;&nbsp;&nbsp;&nbsp; response to a non-INVITE is ever delivered=
 to a UAC.<br>
&gt; Wow (slaps forehead) - very good catch. That's the whole reason the<br=
>
&gt; immediate NOTIFY<br>
&gt; in SIP Events exists.<br>
&gt;<br>
&gt; The consequences of that are going to require some time to think throu=
gh.<br>
<br>
One way to handle this is to:<br>
<br>
- allow multiple Refer-Events-At header fields in the response<br>
- make it the responsibility of the forking proxy to gather the <br>
responses from each fork and consolidate them into a single response.<br>
<br>
This isn't ideal. It requires the proxy to support the feature. And it <br>
means it must special case forking of REFER, since normally it can <br>
simply forward the first 2xx response it gets to a forked request.<br>
<br>
OTOH, I'll guess there is very little current use of out-of-dialog <br>
REFER. So there may not be many backward compatibility issues with that.<br=
>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Thanks,<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paul<br>
<br>
_______________________________________________<br>
sipcore mailing list<br>
sipcore@ietf.org<br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore">https://www.ietf.=
org/mailman/listinfo/sipcore</a><br>
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From nobody Mon Oct  6 16:01:02 2014
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Date: Mon, 06 Oct 2014 19:00:57 -0400
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] New Version Notification for draft-sparks-sipcore-refer-explicit-subscription-01
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On 10/6/14 4:01 PM, Christer Holmberg wrote:
> Hi,
>
> I don't think we shall assume support in forking proxy (why didn't we
> define a Forking-Proxy-Require header field back in the days...).
>
> I suggested to use
> NOTIFY which indicates terminated.

For an in-dialog refer, would that NOTIFY be in the existing dialog? In 
3265 there is a new dialog usage for that, but a goal is to avoid that. 
So if it was in the dialog, then it would have to be in the INVITE 
dialog-usage, which would be a change.

And for out-of-dialog refer, there is no dialog for the notify to travel 
in. Would that NOTIFY be out-of-dialog but non-dialog-establishing? That 
would be a bit weird.

> If that doesn't work, perhaps we
> should disallow the mechanism in out-of-dialog REFERs. After all, the
> need for the mechanism comes from in-dialog usage of REFER...

Perhaps that is possible.

	Thanks,
	Paul

> Regards,
>
> Christer
>
> Sent from my Windows Phone
> ------------------------------------------------------------------------
> From: Paul Kyzivat <mailto:pkyzivat@alum.mit.edu>
> Sent: ý06/ý10/ý2014 22:00
> To: sipcore@ietf.org <mailto:sipcore@ietf.org>
> Subject: Re: [sipcore] New Version Notification for
> draft-sparks-sipcore-refer-explicit-subscription-01
>
> On 10/6/14 10:25 AM, Robert Sparks wrote:
>>
>> On 10/6/14 12:44 AM, OKUMURA Shinji wrote:
>>> Hello Robert,
>>>
>>>> On 10/3/14 10:40 AM, OKUMURA Shinji wrote:
>>>>> Hi,
>>>>>
>>>>> As a general rule, REFER request may fork.
>>>>> But a referrer receives only one response(i.e. Refer-Events-At).
>>>>> Does this draft disallow a REFER request to be forked?
>>>> No.
>>>>
>>>> If the REFER forks, it will get responses from each branch of the fork.
>>>> The response from each branch will have its own information about
>>>> whether
>>>> the REFER was accepted on that branch, including its own Refer-Events-At
>>>> URI.
>>> In accordance with the following, a successful response for REFER
>>> request is only one.
>>>
>>> RFC3261
>>> 17.1.2.1 Overview of the non-INVITE Transaction
>>>
>>>     Unlike an INVITE transaction, a non-INVITE transaction has no special
>>>     handling for the 2xx response.  The result is that only a single 2xx
>>>     response to a non-INVITE is ever delivered to a UAC.
>> Wow (slaps forehead) - very good catch. That's the whole reason the
>> immediate NOTIFY
>> in SIP Events exists.
>>
>> The consequences of that are going to require some time to think through.
>
> One way to handle this is to:
>
> - allow multiple Refer-Events-At header fields in the response
> - make it the responsibility of the forking proxy to gather the
> responses from each fork and consolidate them into a single response.
>
> This isn't ideal. It requires the proxy to support the feature. And it
> means it must special case forking of REFER, since normally it can
> simply forward the first 2xx response it gets to a forked request.
>
> OTOH, I'll guess there is very little current use of out-of-dialog
> REFER. So there may not be many backward compatibility issues with that.
>
>          Thanks,
>          Paul
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From nobody Mon Oct  6 19:30:00 2014
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From: OKUMURA Shinji <ietf.shinji@gmail.com>
To: Brett Tate <brett@broadsoft.com>, sipcore@ietf.org
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Subject: Re: [sipcore] I-D Action: draft-sparks-sipcore-refer-explicit-subscription-00.txt
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Hi,

in the below sequence,

    subscriber                              notifier
        |                                       |
        |   SUBSCRIBE in dialog                 |
        |-------------------------------------->| dialog1
        |   callid:1, fromtag:a, totag:b        |
        |                                       |
        |   200 OK(SUBSCRIBE)                   |
        |<--------------------------------------| dialog1
        |                                       |
        |   NOTIFY                              |
        |<--------------------------------------| dialog2
        |   callid:1, fromtag:c, totag:a        |
        |                                       |

This NOTIFY seems to satisfy the RFC 6665 restriction(section 4.4.1).
Does this NOTIFY act against accepted wisdom for SIP?

>> I just come up with another approach.
>>
>> A referrer send REFER within an existing dialog.
>> And a notifier(referee) send NOTIFY within new dialog.
>> Of course Request URI of NOTIFY is a GRUU of a referrer.
>>
>> It seems to be a simple solution for an avoidance of dialog-reuse.
>>
>> I think this option(Required: subscription-newdialog?) is available
>> for not only REFER but also SUBSCRIBE.
>
>One of the obstacles to this approach is that it would likely require an
>update to RFC 6665.  The NOTIFY is not really a dialog-creating method.
>RFC 6665 also significantly restricted what could create a subscription;
>I'm not sure that this approach would satisfy the RFC 6665 restriction.
>
>-brett


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From: OKUMURA Shinji <ietf.shinji@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Tue, 07 Oct 2014 15:29:49 +0900
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Subject: Re: [sipcore] New Version Notification for draft-sparks-sipcore-refer-explicit-subscription-01
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Hi,

>I suggested to use
>NOTIFY which indicates terminated. If that doesn't work, perhaps we
>should disallow the mechanism in out-of-dialog REFERs. After all,
>the need for the mechanism comes from in-dialog usage of REFER...

Correct me if I'm wrong, would the NOTIFY transport a Refer-Events-At header?

Regards,
Shinji


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Subject: Re: [sipcore] I-D Action: draft-sparks-sipcore-refer-explicit-subscription-00.txt
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>     subscriber                              notifier
>         |                                       |
>         |   SUBSCRIBE in dialog                 |
>         |-------------------------------------->| dialog1
>         |   callid:1, fromtag:a, totag:b        |
>         |                                       |
>         |   200 OK(SUBSCRIBE)                   |
>         |<--------------------------------------| dialog1
>         |                                       |
>         |   NOTIFY                              |
>         |<--------------------------------------| dialog2
>         |   callid:1, fromtag:c, totag:a        |
>         |                                       |
>
> This NOTIFY seems to satisfy the RFC 6665 restriction(section 4.4.1).
> Does this NOTIFY act against accepted wisdom for SIP?

RFC 6665 section 4.2.1.2:

"Upon successfully accepting or refreshing a subscription, notifiers
  MUST send a NOTIFY request immediately to communicate the current
  resource state to the subscriber.  This NOTIFY request is sent on the
  same dialog as created by the SUBSCRIBE response."

RFC 6665 section 4.4.3:

"Upon receipt of this SUBSCRIBE request, the notifier (or notifiers,
  if the SUBSCRIBE request was forked) will send a NOTIFY request
  containing resource state in the same dialog."


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Thread-Topic: [sipcore] New Version Notification for draft-sparks-sipcore-refer-explicit-subscription-01
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Subject: Re: [sipcore] New Version Notification for draft-sparks-sipcore-refer-explicit-subscription-01
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Hi,

>> I don't think we shall assume support in forking proxy (why didn't we=20
>> define a Forking-Proxy-Require header field back in the days...).
>>
>> I suggested to use
>> NOTIFY which indicates terminated.
>
> For an in-dialog refer, would that NOTIFY be in the existing dialog? In
> 3265 there is a new dialog usage for that, but a goal is to avoid that.=20
> So if it was in the dialog, then it would have to be in the INVITE dialog=
-usage, which would be a change.

The NOTIFY would only be sent in the out-of-dialog case. For in-dialog ther=
e is no need for the NOTIFY.

> And for out-of-dialog refer, there is no dialog for the notify to travel =
in. Would that NOTIFY be out-of-dialog but non-dialog-establishing? That wo=
uld be a bit weird.

Correct, when I think about it...

Regards,

Christer




> From: Paul Kyzivat <mailto:pkyzivat@alum.mit.edu>
> Sent: =FD06/=FD10/=FD2014 22:00
> To: sipcore@ietf.org <mailto:sipcore@ietf.org>
> Subject: Re: [sipcore] New Version Notification for
> draft-sparks-sipcore-refer-explicit-subscription-01
>
> On 10/6/14 10:25 AM, Robert Sparks wrote:
>>
>> On 10/6/14 12:44 AM, OKUMURA Shinji wrote:
>>> Hello Robert,
>>>
>>>> On 10/3/14 10:40 AM, OKUMURA Shinji wrote:
>>>>> Hi,
>>>>>
>>>>> As a general rule, REFER request may fork.
>>>>> But a referrer receives only one response(i.e. Refer-Events-At).
>>>>> Does this draft disallow a REFER request to be forked?
>>>> No.
>>>>
>>>> If the REFER forks, it will get responses from each branch of the fork=
.
>>>> The response from each branch will have its own information about=20
>>>> whether the REFER was accepted on that branch, including its own=20
>>>> Refer-Events-At URI.
>>> In accordance with the following, a successful response for REFER=20
>>> request is only one.
>>>
>>> RFC3261
>>> 17.1.2.1 Overview of the non-INVITE Transaction
>>>
>>>     Unlike an INVITE transaction, a non-INVITE transaction has no speci=
al
>>>     handling for the 2xx response.  The result is that only a single 2x=
x
>>>     response to a non-INVITE is ever delivered to a UAC.
>> Wow (slaps forehead) - very good catch. That's the whole reason the=20
>> immediate NOTIFY in SIP Events exists.
>>
>> The consequences of that are going to require some time to think through=
.
>
> One way to handle this is to:
>
> - allow multiple Refer-Events-At header fields in the response
> - make it the responsibility of the forking proxy to gather the=20
> responses from each fork and consolidate them into a single response.
>
> This isn't ideal. It requires the proxy to support the feature. And it=20
> means it must special case forking of REFER, since normally it can=20
> simply forward the first 2xx response it gets to a forked request.
>
> OTOH, I'll guess there is very little current use of out-of-dialog=20
> REFER. So there may not be many backward compatibility issues with that.
>
>          Thanks,
>          Paul
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Robert Sparks <rjsparks@nostrum.com>, OKUMURA Shinji <ietf.shinji@gmail.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: [sipcore] New Version Notification for draft-sparks-sipcore-refer-explicit-subscription-01
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Date: Tue, 7 Oct 2014 11:06:54 +0000
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Subject: Re: [sipcore] New Version Notification for draft-sparks-sipcore-refer-explicit-subscription-01
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Hi,

>>>> As a general rule, REFER request may fork.
>>>> But a referrer receives only one response(i.e. Refer-Events-At).
>>>> Does this draft disallow a REFER request to be forked?
>>> No.
>>>
>>> If the REFER forks, it will get responses from each branch of the fork.
>>> The response from each branch will have its own information about=20
>>> whether the REFER was accepted on that branch, including its own=20
>>> Refer-Events-At URI.
>> In accordance with the following, a successful response for REFER=20
>> request is only one.
>>
>> RFC3261
>> 17.1.2.1 Overview of the non-INVITE Transaction
>>
>>     Unlike an INVITE transaction, a non-INVITE transaction has no specia=
l
>>     handling for the 2xx response.  The result is that only a single 2xx
>>     response to a non-INVITE is ever delivered to a UAC.
>Wow (slaps forehead) - very good catch. That's the whole reason the immedi=
ate NOTIFY in SIP Events exists.
>
>The consequences of that are going to require some time to think through.

I was checking RFC 4488, because you have the same issue there if the immed=
iate NOTIFY is not sent.

RFC 4488 "solves" the problem with the following statement:

	"The "Refer-Sub" header field set to "false" MAY be used by the REFER-
   	Issuer only when the REFER-Issuer can be certain that the REFER
   	request will not be forked."

So, perhaps we could do the same.

Regards.

Christer


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From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [sipcore] A SIP OAuth use case
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--089e0117716fce2ddb0504db4a94
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Hi Christer,

I am not clear on the flow of messages in your case.
Can you please provide a high level diagram that describes that flow?

Also, the token that is returned by the WAF to the SIP endpoint; is it an
access token, as per OAuth 2.0, or a user token (ID Token), as per OpenID?

Thanks,
 Rifaat



On Thu, Sep 25, 2014 at 11:34 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
>
>
> >What I am proposing is something along the lines of OpenID, which
> extends the OAuth to >authenticate the *user*.
>
> >In this case, the mapping between OAuth and SIP would be something like
> the following:
>
> >
>
> >OAuth                                    SIP
>
> >
> ---------------------------------------------------------------------------
>
> >Resource Owner                    End User
>
> >Resource Server                    SIP Proxy or SIP Application Server
>
> >Client                                     SIP UA
>
> >Authorization Server             Authorization Server
>
> >
>
> >Do you see a problem with this model?
>
>
>
> As I indicated earlier, this is a little different mapping than the one
> done by 3GPP.
>
>
>
> *OAuth role                                     IMS role*
>
>
>
> Resource Owner                           IMS subscriber
>
> Client                                               WWSF
>
> Authorization Server                    WAF
>
> Resource Server                            IMS network
>
>
>
> (copied text)
>
>
>
> WWSF (WebRTC Web Server Function)
>
> -          Acts as web server, from where the user downloads the IMS
> access application.
>
> WAF (WebRTC Authorisation Function)
>
> -          Acts as the authorization server (the "facebook"), and issues
> the authorization tokens.
>
>
>
> (The WWSF and WAF can, but does not have to, be located in the same domain
> as the IMS provider.)
>
>
>
> I guess the important thing to note here is that the IMS core network
> (including the registrar/S-CSCF and HSS) does not act as Authorization
> Server, but rather as the Resource Server.
>
>
>
> In fact, the IMS core network itself does not even need to support
> OAuth2.0. Instead the WAF performs mapping and verification between "legacy
> IMS credentials" and "web credentials" (e.g. OAuth2.0). The detailed
> procedures for that mapping/verification is not specified in the current
> 3GPP release.
>
>
>
> Regards,
>
>
>
> Christer
>
>
>
>
>
>
>

--089e0117716fce2ddb0504db4a94
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<div dir=3D"ltr">Hi Christer,<div><br></div><div>I am not clear on the flow=
 of messages in your case.&nbsp;</div><div>Can you please provide a high le=
vel diagram that describes that flow?</div><div><br></div><div>Also, the to=
ken that is returned by the WAF to the SIP endpoint; is it an access token,=
 as per OAuth 2.0, or a user token (ID Token), as per OpenID?</div><div><br=
></div><div>Thanks,</div><div>&nbsp;Rifaat</div><div><br></div><div><br></d=
iv></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Thu, =
Sep 25, 2014 at 11:34 AM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmb=
erg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote"=
 style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">





<div lang=3D"EN-GB" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Hi,<u></u><u></u></span><=
/p>
<div>
<div>
<div><span class=3D"">
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d"><u></u>&nbsp;<u></u></=
span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span>What I am p=
roposing is something along the lines of OpenID, which extends the OAuth to
<span style=3D"color:#1f497d">&gt;</span>authenticate the *user*.<u></u><u>=
</u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span>In this cas=
e, the mapping between OAuth and SIP would be something like the following:=
<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span><u></u>&nbs=
p;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span>OAuth &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nb=
sp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;SIP<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span>-----------=
----------------------------------------------------------------<u></u><u><=
/u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span>Resource Ow=
ner &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;En=
d User<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span>Resource Se=
rver &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;S=
IP Proxy or SIP Application Server<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span>Client &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; SIP UA<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span>Authorizati=
on Server &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Authorization Server<u>=
</u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span><u></u>&nbs=
p;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span>Do you see =
a problem with this model?<u></u><u></u></p>
</div>
</span><div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d"><u></u>&nbsp;<u></u></=
span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">As I indicated earlier, t=
his is a little different mapping than the one done by 3GPP.<u></u><u></u><=
/span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:11.0pt;font-family:&quot=
;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">OAuth role&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; IMS role<u></u><u></=
u></span></b></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Resource Owner&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; IMS =
subscriber<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Client&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; WWSF<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Authorization Server&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; WAF<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Resource Server&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; IMS network<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">(copied text)<u></u><u></=
u></span></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">WWSF (WebRTC Web Server Function)<u></u><u></u></p>
<p><u></u><span>-<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><u></u>Acts as web server, from where the user downloads the =
IMS access application.<u></u><u></u></p>
<p class=3D"MsoNormal">WAF (WebRTC Authorisation Function)<u></u><u></u></p=
>
<p><u></u><span>-<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><u></u>Acts as the authorization server (the &ldquo;facebook&=
rdquo;), and issues the authorization tokens.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">(The WWSF and WAF can, but does not have to, be loca=
ted in the same domain as the IMS provider.)<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">I guess the important thing to note here is that the=
 IMS core network (including the registrar/S-CSCF and HSS) does not act as =
Authorization Server, but rather as the Resource Server.
<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">In fact, the IMS core network itself does not even n=
eed to support OAuth2.0. Instead the WAF performs mapping and verification =
between &ldquo;legacy IMS credentials&rdquo; and &ldquo;web credentials&rdq=
uo; (e.g. OAuth2.0). The detailed procedures for that mapping/verification
 is not specified in the current 3GPP release.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">Regards,<span class=3D"HOEnZb"><font color=3D"#88888=
8"><u></u><u></u></font></span></p><span class=3D"HOEnZb"><font color=3D"#8=
88888">
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<p class=3D"MsoNormal">Christer<u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
</font></span></div>
</div>
</div>
</div>
</div>
</div>

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

--089e0117716fce2ddb0504db4a94--


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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Thread-Topic: [sipcore] A SIP OAuth use case
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Subject: Re: [sipcore] A SIP OAuth use case
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I know I've owed you a response to this one for a while... Still just tryin=
g to understand the requirements.

I suspect that application services in the enterprise, the ones that you wo=
rry about maintaining ACLs, don't get to avoid having "a separate user dire=
ctory" so easily. A classic voicemail service, for example, needs to keep v=
oicemails for Alice separate from Bob - that's why typically it authenticat=
es who Alice and Bob are. Such a service must have effectively a "directory=
" of separate users that effectively implies ACLs.

Now if, as you say:

> The application server needs the following pieces of information about ea=
ch user:
> 1. ID and credentials
> 2. User profile to control the level of service provided to the user.

What is your app server using that ID and credentials for, if not to compar=
e to something that is effectively an ACL - like, to prevent Alice from get=
ting Bob's voicemail? My point earlier was that if all you care about is sh=
owing that someone is Alice or Bob (or just that they're an authenticated u=
ser and it doesn't matter who), then a solution to the requirements could l=
ook very similar to RFC4474 or STIR: you just need authentication, not auth=
orization, and calling it authorization is just confusing the issue.

But, there's a different approach here, one where a service doesn't need to=
 know IDs or credentials or anything about users. This is an approach where=
 I just get a token from an authorization service, and hand that token to t=
he app service, and the app service uses that token *instead of* my ID or c=
redentials or any personal profile about me. I can imagine a very odd voice=
mail service that, rather than putting messages into buckets for Alice and =
Bob, put them all into one generic bucket - and then an outside authorizati=
on service kept track of the fact that messages 2, 17, and 109 are for Alic=
e. When Alice requests access to her voicemail from the authorization servi=
ce, it gives her a token that proves to the voicemail service that she is a=
llowed to access only messages 2, 17, and 109, and she can only do it in th=
e next five minutes, say, to prevent capture and replay. The voicemail serv=
ice then has no idea who she is or who those messages are for, it just poni=
es them up. If *that* is what you said above, at least I would understand w=
hy you want to use OAuth - it would be a terrible architecture (the authZ s=
ervice keeps track of those messages how exactly?), but at least this is wh=
at OAuth is for. The architectures make more sense with third parties.

A third approach would be problems we looked at as "trait-based" authorizat=
ion problems. The classic example is where for some service, perhaps it mat=
ters whether Alice is in the Engineering department or the Finance departme=
nt, rather than who Alice is exactly. Maybe those're the sort of elements y=
ou mean by "user profile." We did a whole requirements process years ago on=
 "trait-based" authorization mechanisms intended to satisfy cases like this=
: see RFC4484. You'll note it shows architectures where a SIP endpoints poi=
nt by-reference to HTTP objects that contain authorization policy elements =
like such "traits." There are a bunch of solutions for that space, too, but=
 if that's what you want, then my question for you would be, do you have an=
y requirements beyond what's defined in RFC4484 Sec. 5?

Jon Peterson
Neustar, Inc.

From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com<mailto:rifaat.ietf@gmail.co=
m>>
Date: Wednesday, September 17, 2014 at 6:22 AM
To: Jon Peterson <jon.peterson@neustar.biz<mailto:jon.peterson@neustar.biz>=
>
Cc: Paul Kyzivat <pkyzivat@alum.mit.edu<mailto:pkyzivat@alum.mit.edu>>, "si=
pcore@ietf.org<mailto:sipcore@ietf.org>" <sipcore@ietf.org<mailto:sipcore@i=
etf.org>>
Subject: Re: [sipcore] A SIP OAuth use case



On Mon, Sep 15, 2014 at 3:29 PM, Peterson, Jon <jon.peterson@neustar.biz<ma=
ilto:jon.peterson@neustar.biz>> wrote:

Format snags aside,

I copied that text from a different editor.
The format seems ok here:
http://www.ietf.org/mail-archive/web/sipcore/current/msg06258.html

So, it might be something on your side?


I have a bit of difficulty even differentiating the issues listed below: su=
rely the reason why enterprises assign multiple identities per user is to a=
uthorize access to different services separately.

Sorry, I do not get that.
Why would the enterprise be required to create multiple identities to allow=
 access to different services separately? in other words, why cannot the en=
terprise create one identity and still be able to control the user's access=
 to different services provided by different servers?


This is an issue that could be resolved with an access control list per ser=
vice and a trusted identity broker in the enterprise vouching for user iden=
tities to these services.

But that means that each server that provides a service must maintain a sep=
arate user directory, which is exactly what I am trying to avoid.


But to convince ourselves that one approach to this is better than another,=
 we need to look at a level of detail below these issue descriptions. Again=
, in order to access a voicemail service in an enterprise, say, what does t=
he voicemail service need to know about the user?

Like, for regular access to the voicemail for "jon@enterprise," surely all =
the voicemail service needs to know is that the entity authoritative for th=
e @enterprise issued me the name "jon." Maybe the "admin" account gets to a=
ccess all the voicemails, if need be.

The application server needs the following pieces of information about each=
 user:
1. ID and credentials
2. User profile to control the level of service provided to the user.

You cannot assume that all users will get the same level of service.


There are a variety of mechanisms that suffice to prove simple identities l=
ike that to a service.

Can you elaborate?
Are you assuming that the service still have a separate user directory?


If, however, I as "jon" needed for a third-party transcription service to a=
ccess a subset of the messages in my voicemail box, and I wanted the permis=
sions for that access to expire as soon as the messages had been transcribe=
d, then there is a much narrower set of mechanisms that could provide that =
capability.

You keep going back to the OAuth 2.0 model and try to impose it on SIP, whi=
ch would work in some use cases.

What I am proposing is something along the lines of OpenID, which extends t=
he OAuth to authenticate the *user*.
In this case, the mapping between OAuth and SIP would be something like the=
 following:

OAuth                                    SIP
---------------------------------------------------------------------------
Resource Owner                     End User
Resource Server                     SIP Proxy or SIP Application Server
Client                                     SIP UA
Authorization Server                Authorization Server

Do you see a problem with this model?


But it sounds to me, from this issues list, like your requirements are more=
 along the former lines than the latter.


Actually, it is both.

Here is an example of a use case that fits with the OAuth 2.0 model:

An Application Server that provides Mobile Clients with access to SIP servi=
ces.
The Mobile Client uses a proprietary protocol to communicate with the Appli=
cation Server that registers and gets service from the SIP network on behal=
f of the user that is using the Mobile Client.

The communication channels between the players in this case are as follows =
(hopefully the format is ok this time):

[Mobile Client]  <----------->  [Application Server] <-----------> [SIP Pro=
xy]
          /\                                         /\
           |                                          |
           |                                         \/
           ------------------------> [Authorization Server]


In this case, the mapping between OAuth and SIP would be something like the=
 following:

OAuth                                    SIP
---------------------------------------------------------------------------
Resource Owner                     End User
Resource Server                     SIP Proxy
Client                                     Application Server (providing se=
rvice to the mobile client)
Authorization Server                Authorization Server


Regards,
 Rifaat




Jon Peterson
Neustar, Inc.

From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com<mailto:rifaat.ietf@gmail.co=
m>>
Date: Saturday, September 13, 2014 at 12:32 PM

To: Jon Peterson <jon.peterson@neustar.biz<mailto:jon.peterson@neustar.biz>=
>
Cc: Paul Kyzivat <pkyzivat@alum.mit.edu<mailto:pkyzivat@alum.mit.edu>>, "si=
pcore@ietf.org<mailto:sipcore@ietf.org>" <sipcore@ietf.org<mailto:sipcore@i=
etf.org>>
Subject: Re: [sipcore] A SIP OAuth use case



On Thu, Sep 4, 2014 at 7:27 PM, Peterson, Jon <jon.peterson@neustar.biz<mai=
lto:jon.peterson@neustar.biz>> wrote:

I understand that OAuth can be altered from its basic flows to provide toke=
ns for OpenID, and IdP could be bound to OAuth, and while this is all very =
interesting it doesn't change much about the requirements discussion I thin=
k we need to have: the one about what kinds of use cases we want to satisfy=
 and who the relying parties are in those cases. Once we understand that, w=
e will understand what tools are applicable to these use cases - preferably=
, not tools that need to be twisted into a different shape to be applicable=
.


Well, OpenID extends the OAuth 2.0 Framework to authenticate the user; why =
do you think it is "twisting" OAuth into a different shape?



It would helpful, in the enterprise SSO case, for us to understand what kin=
ds of application servers a user might need to access, and what exactly tho=
se application servers need to know about the user in order to accomplish t=
heir function.


Below is a description for some enterprise issues, that hopefully will be a=
ddressed by a new authorization framework:


* Multiple Identities One of the major issues in the enterprise is the need=
 for a user to obtain multiple uncorrelated identities associated with diff=
erent products to get access to the various services provided by the enterp=
rise. For example, in the enterprise the user will be provided with differe=
nt identities to: o Login his device (PC, laptop, mobile, ...) to the netwo=
rk. o Register to the SIP network and get basic SIP services. o Access his =
Voice Mail. o Host a conference. o etc Typically, these identities belong t=
o different administrative domains and realms. Instead of these multiple id=
entities, ideally the enterprise would like to create and maintain only one=
 identity for the user and then allow the user access to all SIP and non-SI=
P services using that one identity. * Authorization of Access to Services A=
nother issue in the enterprise is the need to control and authorize a user =
access to services. Because application servers have their own user directo=
ry, the control of access to services is spread over a multitude of servers=
 in the network. Ideally, the enterprise would like to manage the access to=
 services using one centralized entity and for the various entities that pr=
ovide the service to become the enforcement points. * Assignment of Service=
s In the enterprise, different users might be assigned to different servers=
 providing a specific service; e.g. Voice Mail Box for different users migh=
t be hosted by different servers, and the user need to know his assigned se=
rver to be able to get access to this service. * Application Servers To be =
able to provide service, the application server needs a list of users allow=
ed to get services. The application server needs the following pieces of in=
formation about each user: 1. User ID and Credentials 2. Profile to control=
 the level of service provided to the user. If one application server provi=
des different services using different protocols, the application server mi=
ght need to maintain different identities associated with these protocols; =
e.g. an application server that provides presence service using SIP, and IM=
 service using XMPP. Ideally, an authorization framework would: 1. Off load=
 the application server from the need to manage the user credentials and au=
thenticating the user. 2. Off load the application server from the need to =
maintain the users and their profiles to control the level of service provi=
ded to the user. 3. Allow the assignment of a user to a specific applicatio=
n server. * OAuth/OpenID Model With the OAuth/OpenID framework, the user ID=
 and credentials will be handled by the authorization server, and the appli=
cation server assignment and the level of service associated with the user =
will be provided in the scope of the token associated with the user. When t=
he application server receives a request from a user it will have all the d=
etails that would allow the application server to validate the request to m=
ake sure it is coming from the right authorization server, and to act upon =
the request because it has the scope associated with the user, without the =
need to contact the authorization server. * Relying Party Regarding the Rel=
ying Party question: I think that when the user is being authenticated, tha=
t the Relying Party would be the SIP Client that would be given a user toke=
n (ID Token in OpenID lingo) to access services on behalf of the end user. =
So, the mapping would be something like this: +------------------------+---=
--------------------------+ | OAuth | SIP | +------------------------+-----=
------------------------+ | Resource Owner | End User | | Resource Server |=
 SIP Proxy or SIP App Server | | Client | SIP UA (Relying Party) | | Author=
ization Server | Authorization Server | +------------------------+---------=
--------------------+


Regards,
 Rifaat






Jon Peterson
Neustar, Inc.

From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com<mailto:rifaat.ietf@gmail.co=
m>>
Date: Thursday, September 4, 2014 at 8:31 AM
To: Jon Peterson <jon.peterson@neustar.biz<mailto:jon.peterson@neustar.biz>=
>
Cc: Paul Kyzivat <pkyzivat@alum.mit.edu<mailto:pkyzivat@alum.mit.edu>>, "si=
pcore@ietf.org<mailto:sipcore@ietf.org>" <sipcore@ietf.org<mailto:sipcore@i=
etf.org>>
Subject: Re: [sipcore] A SIP OAuth use case




On Wed, Sep 3, 2014 at 4:00 PM, Peterson, Jon <jon.peterson@neustar.biz<mai=
lto:jon.peterson@neustar.biz>> wrote:

While I would echo Paul's thoughts below about the plausibility of service
providers using this model to authorize recording, I also agree that
third-party recording is an excellent example of a service architecture
that requires an authorization mechanism like OAuth. An end user needs to
authorize a third party to participate in the session under certain
constraints, and coordinate the association between the third party and
the VoIP service. This example also (rightly, in my opinion) conducts the
majority of the flow in HTTP where it belongs.


I agree with that.


Many of the use cases under discussion in this thread, though, are of the
form where there are really only two involved parties: an end user and a
service provider. If you are my service provider, then in traditional SIP
I share a secret with you that I use to REGISTER my devices. I don't think
you need OAuth for that.

If you assume that there is one server that provides all the services, then=
 you are right.
But that is not the case all the time. I think that Dale articulated that c=
learly in the following response:
http://www.ietf.org/mail-archive/web/sipcore/current/msg06233.html


I do understand that OAuth 2.0 is used in the
wild for SSO, though I sympathize with Matt that the base flow shown in
RFC6749 Sec 1.2 just doesn't map onto the requirements for SSO.


I agree that the OAuth 2.0 "as is" does not address the SSO requirements.
The SIP OAuth proposal is not just using OAuth 2.0 "as is", but it is exten=
ding the framework to authenticate and provide a *user* specific token.

Since Matt mentioned OpenID few times, I looked at the OpenID specification=
.
What they are doing there is that they defined a new token, called ID Token=
, that is associated with the *user*; which is exactly what the SIP OAuth p=
roposal is doing.
http://openid.net/specs/openid-connect-core-1_0.html#IDToken

My plan is to use the same terminology, ID Token, in the next version of th=
e SIP OAuth document to clarify that, and to differentiate it from the acce=
ss token.


I think, though, we might usefully break uses cases here down by who the
intended relying party is:

- The user's own SIP service provider (my enterprise, my carrier, my OTT
provider)
- A third party that a user wants to authorize for a service (recording
service... But anything else?)

I would also append some use cases where the relying party:

- Someone with no association with the user (caller ID sorts of cases)

This last is where I've argued things like IdP might be relevant.


EKR has a nice description for binding IdP to OAuth in his WebRTC Security =
Architecture document here:
https://tools.ietf.org/html/draft-ietf-rtcweb-security-arch-10#appendix-A.2

We could use the same idea for SIP.


Regards,
 Rifaat



Are there any other high-level buckets of use cases here?

Jon Peterson
Neustar, Inc.

On 8/25/14, 11:44 AM, "Paul Kyzivat" <pkyzivat@alum.mit.edu<mailto:pkyzivat=
@alum.mit.edu>> wrote:

>On 8/25/14 11:50 AM, Rifaat Shekh-Yusef wrote:
>> Hi Matt,
>>
>> The use case and the solution provided assumes that the VoIP Provider
>> makes the decision on when and what to record.
>> While this is a valid use case, there are other use cases that allow the
>> user to decide on when and what to record; take a look at RFC6341 for
>> more info.
>> http://datatracker.ietf.org/doc/rfc6341/
>
>SIPREC supports numerous topologies.
>It would support the one that Matt outlines as well as end-user
>controlled ones.
>
>I found Matt's use case interesting. I think it would be nice if such
>services were available in that form. But I have difficulty imagining
>any of the VoIP providers I'm aware of supporting this model. My
>impression is that this means that they are giving up a value added
>revenue market, that they are loath to do. The most likely justification
>I can imagine that they don't want to legal consequences of doing
>recording.
>
>       Thanks,
>       Paul
>
>> For these use cases, where the user is in control, you need to away to
>> authenticate and control which users are allowed to record and the level
>> of service provided for that specific user.
>> This is where the solutions provided in section 3 or section 4 of the
>> SIP OAuth proposal come into play.
>> Obviously, in this case the authorization server is different from
>> authorization server used to establish the trust relationship between
>> the VoIP Provider and the CRS Provider.
>> In this case, the authorization server must authenticate the user and
>> provide his UA with an access token or a code that controls the user's
>> access to the service.
>>
>> Regards,
>>   Rifaat
>>
>>
>>
>> On Sat, Aug 23, 2014 at 9:05 AM, Matt Ryan <ietf@mvryan.org<mailto:ietf@=
mvryan.org>
>> <mailto:ietf@mvryan.org<mailto:ietf@mvryan.org>>> wrote:
>>
>>     Included is the use case I spoke of before.  My apologies for the
>>delay.
>>
>>     This is written as though it could be included in
>>     [draft-yusef-sipcore-sip-oauth] at section 6.
>>
>>
>>     6. Examples
>>
>>     6.1. Example: Call Recording Service
>>
>>     This example illustrates the use of SIP OAuth between three
>>     parties:  A hosted VoIP service provider, a Call Recording Service
>>     provider, and a phone system end-user.  In this example:
>>     - The phone system end-user is a customer of both the hosted VoIP
>>     provider and the Call Recording Service provider
>>     - The hosted VoIP service provider is a standard provider of hosted
>>     business-class VoIP telephone service using SIP
>>     - The Call Recording Service provider is a distinct entity from the
>>     hosted VoIP provider
>>
>>     The Call Recording Service provides to customers both the call
>>     recording abilities and management of call recordings
>>     (configuration, sharing and distribution, retention, etc.).  This
>>     service can accept and record RTP traffic, associate all RTP streams
>>     with a single call together, and store and catalog the recorded data
>>     for future searching and retrieval.  The service also offers a web
>>     interface where customers may view and retrieve stored calls.
>>     Stored calls may range from simple person-to-person voice calls to
>>     multi-party conferences with a multitude of audio and video
>>     streams.  In this example, customers are billed based on the amount
>>     of data they store, although other models are certainly possible.
>>
>>     6.1.1. OAuth 2.0 Roles
>>
>>     In this example, each party assumes the following OAuth 2.0 roles as
>>     defined in [RFC6749] Section 1.1:
>>     - The end-user assumes the role of resource owner
>>     - The hosted VoIP service provider assumes the role of client
>>     - The Call Recording Service provider assumes the role of resource
>>     server as well as the role of authorization server
>>
>>     6.1.2. Description
>>
>>     There are two parts to the example:  Initial configuration and
>>     real-time recording authorization.
>>
>>     Each section includes a simplified flow diagram for the purpose of
>>     describing the corresponding portion of the example.  For the sake
>>     of brevity, certain parts of the OAuth dialog have been eliminated.
>>     Implements should refer to [RFC6749] for details on OAuth.
>>
>>     6.1.2.1 Initial Configuration
>>
>>        +-------------+     +---------------+     +--------------+
>>        |  End-User   |     | VoIP Provider |     | CRS Provider |
>>        | Web Browser |     |    Website    |     |              |
>>        +-------------+     +---------------+     +--------------+
>>               |                    |                     |
>>               | HTTP 1             |                     |
>>               | (Configure CRS)    |                     |
>>               |------------------->|                     |
>>               |                    |                     |
>>               | HTTP 2             |                     |
>>               | (OAuth Redirect    |                     |
>>               |  to CRS Website)   |                     |
>>               |<-------------------|                     |
>>               |                    |                     |
>>               | HTTP 3             |                     |
>>               | (Authorize SIP from VoIP provider        |
>>               |----------------------------------------->|
>>               |                    |                     |
>>               | HTTP 4             |                     |
>>               | (OAuth Redirect back to VoIP portal)     |
>>               |<-----------------------------------------|
>>               |                    |                     |
>>               | HTTP 5             |                     |
>>               |------------------->|                     |
>>               |                    | HTTP 6              |
>>               |                    | (Request Access and |
>>               |                    |  refresh tokens)    |
>>               |                    |-------------------->|
>>               |                    |                     |
>>
>>     Some time after having signed up for both services, but prior to
>>     attempting to record any calls, the end-user logs into the web
>>     portal of the hosted VoIP provider with a web browser and configures
>>     their call recording service (HTTP 1).  This configuration includes
>>     providing a URI where the call recording service may be reached,
>>     along with a set of API credentials, provided by the call recording
>>     service, which will allow the client to initiate an OAuth exchange.
>>     The end-user also configures the conditions under which call
>>     recordings should take place.  For example, the end-user may request
>>     to record every multi-party (conference) call, or every call
>>     involving a specific set of users.  The end-user may also specify
>>     other restrictions, such as restricting codec negotiation to G.711u
>>     for audio and H.264 for video in order to record the RTP streams.
>>     Once the end-user saves the configuration, the hosted VoIP provider
>>     web portal redirects the end-user's web browser to the call
>>     recording service website and a standard HTTP OAuth dialog begins
>>     (HTTP 2).  The end-user authorizes the hosted VoIP provider to
>>     access (i.e. send SIP and RTP traffic to) the call recording
>>     service, for the purpose of recording calls, so long as the
>>     configured conditions are met (HTTP 3).  The result of this
>>     interaction is that the hosted VoIP provider ends up receiving an
>>     OAuth access token and refresh token from the call recording service
>>     (HTTP 4, HTTP 5, HTTP 6).  These tokens are saved in the end-user's
>>     hosted VoIP account database.
>>
>>     6.1.2.2 Real-time Recording Authorization
>>
>>        +-------------+     +---------------+      +--------------+
>>        |  End-User   |     | VoIP Provider |      | CRS Provider |
>>        |  SIP Phone  |     |    Website    |      |              |
>>        +-------------+     +---------------+      +--------------+
>>               |                    |                      |
>>               | SIP INVITE 1       |                      |
>>               |------------------->|                      |
>>               |                    | SIP INVITE 2         |
>>               |                    | (with access token)  |
>>               |                    |--------------------->|
>>               |                    |                      |
>>               |                    | SIP 401 Unauthorized |
>>               |                    |<---------------------|
>>               |                    |                      |
>>               |                    | SIP INVITE 3         |
>>               |                    | (with refresh token) |
>>               |                    |--------------------->|
>>               |                    |                      |
>>               |                    | SIP 200 1            |
>>               |                    | (new access token)   |
>>               |                    |<---------------------|
>>               |                    |                      |
>>               |                    | SIP INVITE 4         |
>>               |                    | (with access token)  |
>>               |                    |--------------------->|
>>               |                    |                      |
>>               |                    | SIP 200 2            |
>>               |                    |<---------------------|
>>               |                    |                      |
>>
>>     Independently of and after the initial configuration phase, the
>>     end-user makes a telephone call using the hosted VoIP provider (SIP
>>     INVITE 1).  When this call takes place, the hosted VoIP provider
>>     looks to see whether it believes this call should be recorded.  If
>>     so, at this point it would branch the call session to the call
>>     recording service, using SIP OAuth to pass the previously provided
>>     access token for authorization (SIP INVITE 2).  Once the access
>>     token is validated by the call recording service (perhaps after
>>     first providing a new access token based on receipt of a valid
>>     refresh token), the call recording service will check the rights
>>     that were previously authorized and examine the SIP to determine
>>     whether it can accept the call.  If so, it completes the
>>     establishment of the session and begins receiving and recording the
>>     RTP stream(s) (SIP 200 OK).  The call recording service provider
>>     could also deny the request, either because the tokens are invalid
>>     or because the content of
>>       the SIP invite does not match the previously authorized
>>     conditions, in which case a SIP 403 would be returned by the call
>>     recording service provider and the call would not be recorded -
>>     however, the initial call branch would continue without
>>interruption.
>>
>>     Note:
>>     [RFC6749] specifies that when a client uses a refresh token to
>>     request a new access token, the response is HTTP 200 with the new
>>     access token and optionally a new refresh token included in the
>>     payload.  In this example, a SIP 200 response (SIP 200 1) is sent
>>     which contains the new token(s).  However, this could be confusing
>>     as SIP 200 is generally viewed as the acceptance of the INVITE,
>>     which is not what is happening in this case.  Should SIP 200 be used
>>     for consistency, or should another mechanism be used?
>>
>>     6.1.3. Justification
>>
>>     6.1.3.1. Hosted VoIP Service Provider
>>
>>     In this example, the hosted VoIP service provider can allow their
>>     customers to enable call recording in their VoIP service by
>>     selecting from any of a number of supported call recording service
>>     providers.  This allows the hosted VoIP service provider to offer
>>     this feature to customers without requiring that the hosted VoIP
>>     service provider implement and support this feature themselves.
>>
>>     6.1.3.2. Call Recording Service Provider
>>
>>     In this example, the Call Recording Service provider can focus on
>>     value and innovation in the area of recording calls and managing
>>     recorded calls without having to build a full-featured telephony
>>     offering.
>>
>>     6.1.3.3. Customer
>>
>>     In this example, the customer has more flexibility in choosing a
>>     complete solution that meets all of the customer needs.  The
>>     customer does not have to settle for a substandard call recording
>>     service offering in order to obtain other features they seek in a
>>     hosted VoIP provider, and vice versa.
>>
>>     6.1.4. Variants
>>
>>     A simple variant of this example is one wherein one of the services
>>     (either the VoIP service or the call recording service) is managed
>>     directly by the end-user, but the other is not.  For example, the
>>     end-user may wish to make use of an external hosted VoIP service
>>     provider, but may have business or legal reasons that forbid the
>>     end-user from allowing a third party to store and manage recorded
>>     calls.  The end-user could choose to set up their own call recording
>>     service as described in this example, and use SIP OAuth to
>>     facilitate interaction between the hosted VoIP service and the
>>     end-user's own call recording service.
>>
>>     6.2. Other SIP OAuth examples
>>
>>     Many other SIP-based services could leverage SIP OAuth to provide
>>     value-added service to an end-user of a hosted VoIP service
>>     provider.  Some examples of these types of services are listed
>>below.
>>
>>     Voicemail service provider:  The end-user configures their hosted
>>     VoIP service provider to manage voicemail through a separate
>>     voicemail service provider, which chooses whether to store voicemail
>>     based on existing quotas, Caller ID, etc.
>>
>>     Conferencing service provider:  A conferencing service may wish to
>>     bill customers in a more granular fashion, for example, based on the
>>     number of participants and attendees in a conference, the duration
>>     of conference, whether video was also included, which codecs were
>>     being used, etc. instead of a flat monthly rate.  SIP OAuth would
>>     facilitate metered authorization for a client's use of the service,
>>     instead of all-or-nothing access.
>>
>>     Call center service provider:  A third party may provide ring groups
>>     or call queues for a hosted VoIP provider along with detailed
>>     reports and dashboards.  The end-user uses OAuth over SIP to control
>>     who can log into which queues or ring groups, etc.
>>
>>
>>
>>     --
>>     Matt Ryan
>>     code slinger | zoomulus.com<http://zoomulus.com> <http://zoomulus.co=
m>
>>     ietf at mvryan dot org
>>
>>
>>
>>
>>     _______________________________________________
>>     sipcore mailing list
>>     sipcore@ietf.org<mailto:sipcore@ietf.org> <mailto:sipcore@ietf.org<m=
ailto:sipcore@ietf.org>>
>>     https://www.ietf.org/mailman/listinfo/sipcore
>>
>>
>>
>>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org<mailto:sipcore@ietf.org>
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
>
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--_000_D059A86A134723jonpetersonneustarbiz_
Content-Type: text/html; charset="us-ascii"
Content-ID: <16D85952C4D8434CBB1673A7C951F889@neustar.biz>
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; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif;">
<div><br>
</div>
<div>I know I've owed you a response to this one for a while... Still just =
trying to understand the requirements.</div>
<div><br>
</div>
<div>I suspect that application services in the enterprise, the ones that y=
ou worry about maintaining ACLs, don't get to avoid having &quot;a separate=
 user directory&quot; so easily. A classic voicemail service, for example, =
needs to keep voicemails for Alice separate
 from Bob - that's why typically it authenticates who Alice and Bob are. Su=
ch a service must have effectively a &quot;directory&quot; of separate user=
s that effectively implies ACLs.</div>
<div><br>
</div>
<div>Now if, as you say:</div>
<div><br>
</div>
<div>
<div>&gt; The application server needs the following pieces of information =
about each user:</div>
<div>&gt; 1. ID and credentials</div>
<div>&gt; 2. User profile to control the level of service provided to the u=
ser.</div>
</div>
<div><br>
</div>
<div>What is your app server using that ID and credentials for, if not to c=
ompare to something that is effectively an ACL - like, to prevent Alice fro=
m getting Bob's voicemail? My point earlier was that if all you care about =
is showing that someone is Alice
 or Bob (or just that they're an authenticated user and it doesn't matter w=
ho), then a solution to the requirements could look very similar to RFC4474=
 or STIR: you just need authentication, not authorization, and calling it a=
uthorization is just confusing the
 issue.</div>
<div><br>
</div>
<div>But, there's a different approach here, one where a service doesn't ne=
ed to know IDs or credentials or anything about users. This is an approach =
where I just get a token from an authorization service, and hand that token=
 to the app service, and the app
 service uses that token *instead of* my ID or credentials or any personal =
profile about me. I can imagine a very odd voicemail service that, rather t=
han putting messages into buckets for Alice and Bob, put them all into one =
generic bucket - and then an outside
 authorization service kept track of the fact that messages 2, 17, and 109 =
are for Alice. When Alice requests access to her voicemail from the authori=
zation service, it gives her a token that proves to the voicemail service t=
hat she is allowed to access only
 messages 2, 17, and 109, and she can only do it in the next five minutes, =
say, to prevent capture and replay. The voicemail service then has no idea =
who she is or who those messages are for, it just ponies them up. If *that*=
 is what you said above, at least
 I would understand why you want to use OAuth - it would be a terrible arch=
itecture (the authZ service keeps track of those messages how exactly?), bu=
t at least this is what OAuth is for. The architectures make more sense wit=
h third parties.</div>
<div><br>
</div>
<div>A third approach would be problems we looked at as &quot;trait-based&q=
uot; authorization problems. The classic example is where for some service,=
 perhaps it matters whether Alice is in the Engineering department or the F=
inance department, rather than who Alice is
 exactly. Maybe those're the sort of elements you mean by &quot;user profil=
e.&quot; We did a whole requirements process years ago on &quot;trait-based=
&quot; authorization mechanisms intended to satisfy cases like this: see RF=
C4484. You'll note it shows architectures where a SIP
 endpoints point by-reference to HTTP objects that contain authorization po=
licy elements like such &quot;traits.&quot; There are a bunch of solutions =
for that space, too, but if that's what you want, then my question for you =
would be, do you have any requirements beyond
 what's defined in RFC4484 Sec. 5?</div>
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div style=3D"font-family:Calibri; font-size:11pt; text-align:left; color:b=
lack; BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM:=
 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid;=
 BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<span style=3D"font-weight:bold">From: </span>Rifaat Shekh-Yusef &lt;<a hre=
f=3D"mailto:rifaat.ietf@gmail.com">rifaat.ietf@gmail.com</a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Wednesday, September 17, 2014=
 at 6:22 AM<br>
<span style=3D"font-weight:bold">To: </span>Jon Peterson &lt;<a href=3D"mai=
lto:jon.peterson@neustar.biz">jon.peterson@neustar.biz</a>&gt;<br>
<span style=3D"font-weight:bold">Cc: </span>Paul Kyzivat &lt;<a href=3D"mai=
lto:pkyzivat@alum.mit.edu">pkyzivat@alum.mit.edu</a>&gt;, &quot;<a href=3D"=
mailto:sipcore@ietf.org">sipcore@ietf.org</a>&quot; &lt;<a href=3D"mailto:s=
ipcore@ietf.org">sipcore@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [sipcore] A SIP OAuth =
use case<br>
</div>
<div><br>
</div>
<div>
<div>
<div dir=3D"ltr"><br>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Mon, Sep 15, 2014 at 3:29 PM, Peterson, Jon <=
span dir=3D"ltr">
&lt;<a href=3D"mailto:jon.peterson@neustar.biz" target=3D"_blank">jon.peter=
son@neustar.biz</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div><br>
</div>
<div>Format snags aside, </div>
</div>
</blockquote>
<div><br>
</div>
<div>I copied that text from a different editor.</div>
<div>The format seems ok here:</div>
<div><a href=3D"http://www.ietf.org/mail-archive/web/sipcore/current/msg062=
58.html" target=3D"_blank">http://www.ietf.org/mail-archive/web/sipcore/cur=
rent/msg06258.html</a><br>
</div>
<div><br>
</div>
<div>So, it might be something on your side?</div>
<div><br>
</div>
<div>&nbsp;<br>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div>I have a bit of difficulty even differentiating the issues listed belo=
w: surely the reason why enterprises assign multiple identities per user is=
 to authorize access to different services separately.
</div>
</div>
</blockquote>
<div><br>
</div>
<div>Sorry, I do not get that.&nbsp;</div>
<div>Why would the enterprise be required to create multiple identities to =
allow access to different services separately? in other words, why cannot t=
he enterprise create one identity and still be able to control the user's a=
ccess to different services provided
 by different servers?<br>
</div>
<div><br>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div>This is an issue that could be resolved with an access control list pe=
r service and a trusted identity broker in the enterprise vouching for user=
 identities to these services.
</div>
</div>
</blockquote>
<div><br>
</div>
<div>But that means that each server that provides a service must maintain =
a separate user directory, which is exactly what I am trying to avoid.</div=
>
<div><br>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div>But to convince ourselves that one approach to this is better than ano=
ther, we need to look at a level of detail below these issue descriptions. =
Again, in order to access a voicemail service in an enterprise, say, what d=
oes the voicemail service need to
 know about the user?</div>
<div><br>
</div>
<div>Like, for regular access to the voicemail for &quot;jon@enterprise,&qu=
ot; surely all the voicemail service needs to know is that the entity autho=
ritative for the @enterprise issued me the name &quot;jon.&quot; Maybe the =
&quot;admin&quot; account gets to access all the voicemails, if
 need be. </div>
</div>
</blockquote>
<div><br>
</div>
<div>
<div>The application server needs the following pieces of information about=
 each user:</div>
<div>1. ID and credentials</div>
<div>2. User profile to control the level of service provided to the user.<=
/div>
</div>
<div><br>
</div>
<div>You cannot assume that all users will get the same level of service.</=
div>
<div><br>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div>There are a variety of mechanisms that suffice to prove simple identit=
ies like that to a service.
</div>
</div>
</blockquote>
<div><br>
</div>
<div>Can you elaborate?</div>
<div>Are you assuming that the service still have a separate user directory=
?</div>
<div><br>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div>If, however, I as &quot;jon&quot; needed for a third-party transcripti=
on service to access a subset of the messages in my voicemail box, and I wa=
nted the permissions for that access to expire as soon as the messages had =
been transcribed, then there is a much narrower
 set of mechanisms that could provide that capability. </div>
</div>
</blockquote>
<div><br>
</div>
<div>You keep going back to the OAuth 2.0 model and try to impose it on SIP=
, which would work in some use cases.</div>
<div><br>
</div>
<div>What I am proposing is something along the lines of OpenID, which exte=
nds the OAuth to authenticate the *user*.</div>
<div>In this case, the mapping between OAuth and SIP would be something lik=
e the following:</div>
<div><br>
</div>
<div>OAuth &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;SIP</div>
<div>----------------------------------------------------------------------=
-----</div>
<div>Resource Owner &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=
 &nbsp; &nbsp; End User</div>
<div>Resource Server &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
; &nbsp; &nbsp; SIP Proxy or SIP Application Server</div>
<div>Client &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; SIP UA</div>
<div>Authorization Server &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;Authorization Server</div>
<div><br>
</div>
<div>Do you see a problem with this model?</div>
<div><br>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div>But it sounds to me, from this issues list, like your requirements are=
 more along the former lines than the latter.&nbsp;</div>
<div><br>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>Actually, it is both.</div>
<div><br>
</div>
<div>Here is an example of a use case that fits with the OAuth 2.0 model:</=
div>
<div><br>
</div>
<div>An Application Server that provides Mobile Clients with access to SIP =
services.</div>
<div>The Mobile Client uses a proprietary protocol to communicate with the =
Application Server that registers and gets service from the SIP network on =
behalf of the user that is using the Mobile Client.</div>
<div><br>
</div>
<div>The communication channels between the players in this case are as fol=
lows (hopefully the format is ok this time):</div>
<div><br>
</div>
<div>[Mobile Client] &nbsp;&lt;-----------&gt; &nbsp;[Application Server] &=
lt;-----------&gt; [SIP Proxy]</div>
<div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; /\ &nbsp; &nbsp; &nbsp; &nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp; /\</div>
<div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp;=
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|</div>
<div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp;=
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; \/<br>
</div>
<div>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;------------------------&gt; =
[Authorization Server]</div>
<div><br>
</div>
<div><br>
</div>
<div>In this case, the mapping between OAuth and SIP would be something lik=
e the following:<br>
</div>
<div>
<div><br>
</div>
<div>OAuth &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;SIP</div>
<div>----------------------------------------------------------------------=
-----</div>
<div>Resource Owner &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=
 &nbsp; &nbsp; End User</div>
<div>Resource Server &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp=
; &nbsp; &nbsp; SIP Proxy</div>
<div>Client &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Application =
Server (providing service to the mobile client)</div>
<div>Authorization Server &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;Authorization Server</div>
</div>
<div><br>
</div>
<div><br>
</div>
<div>Regards,</div>
<div>&nbsp;Rifaat</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div></div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
<div><br>
</div>
<span>
<div style=3D"font-family:Calibri;font-size:11pt;text-align:left;color:blac=
k;border-width:1pt medium medium;border-style:solid none none;padding:3pt 0=
in 0in;border-top-color:rgb(181,196,223)">
<span style=3D"font-weight:bold">From: </span>Rifaat Shekh-Yusef &lt;<a hre=
f=3D"mailto:rifaat.ietf@gmail.com" target=3D"_blank">rifaat.ietf@gmail.com<=
/a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Saturday, September 13, 2014 =
at 12:32 PM
<div>
<div><br>
<span style=3D"font-weight:bold">To: </span>Jon Peterson &lt;<a href=3D"mai=
lto:jon.peterson@neustar.biz" target=3D"_blank">jon.peterson@neustar.biz</a=
>&gt;<br>
<span style=3D"font-weight:bold">Cc: </span>Paul Kyzivat &lt;<a href=3D"mai=
lto:pkyzivat@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit.edu</a>&gt;,=
 &quot;<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.o=
rg</a>&quot; &lt;<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipc=
ore@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [sipcore] A SIP OAuth =
use case<br>
</div>
</div>
</div>
<div>
<div>
<div><br>
</div>
<div>
<div>
<div dir=3D"ltr"><br>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Thu, Sep 4, 2014 at 7:27 PM, Peterson, Jon <s=
pan dir=3D"ltr">
&lt;<a href=3D"mailto:jon.peterson@neustar.biz" target=3D"_blank">jon.peter=
son@neustar.biz</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div><br>
</div>
<div>I understand that OAuth can be altered from its basic flows to provide=
 tokens for OpenID, and IdP could be bound to OAuth, and while this is all =
very interesting it doesn't change much about the requirements discussion I=
 think we need to have: the one
 about what kinds of use cases we want to satisfy and who the relying parti=
es are in those cases. Once we understand that, we will understand what too=
ls are applicable to these use cases - preferably, not tools that need to b=
e twisted into a different shape
 to be applicable.</div>
<div><br>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>Well, OpenID extends the OAuth 2.0 Framework to authenticate the user;=
 why do you think it is &quot;twisting&quot; OAuth into a different shape?<=
/div>
<div><br>
</div>
<div><br>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div></div>
<div>It would helpful, in the enterprise SSO case, for us to understand wha=
t kinds of application servers a user might need to access, and what exactl=
y those application servers need to know about the user in order to accompl=
ish their function.</div>
<div><br>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>Below is a description for some enterprise issues, that hopefully will=
 be addressed by a new authorization framework:</div>
<div><br>
</div>
<div><br>
</div>
<div><span style=3D"color: rgb(0, 0, 0); font-family: 'Courier New', Courie=
r, monospace; font-size: 14px; white-space: pre-wrap;">* Multiple Identitie=
s One of the major issues in the enterprise is the need for a user to obtai=
n multiple uncorrelated identities
 associated with different products to get access to the various services p=
rovided by the enterprise. For example, in the enterprise the user will be =
provided with different identities to: o Login his device (PC, laptop, mobi=
le, ...) to the network. o Register
 to the SIP network and get basic SIP services. o Access his Voice Mail. o =
Host a conference. o etc Typically, these identities belong to different ad=
ministrative domains and realms. Instead of these multiple identities, idea=
lly the enterprise would like to
 create and maintain only one identity for the user and then allow the user=
 access to all SIP and non-SIP services using that one identity. * Authoriz=
ation of Access to Services Another issue in the enterprise is the need to =
control and authorize a user access
 to services. Because application servers have their own user directory, th=
e control of access to services is spread over a multitude of servers in th=
e network. Ideally, the enterprise would like to manage the access to servi=
ces using one centralized entity
 and for the various entities that provide the service to become the enforc=
ement points. * Assignment of Services In the enterprise, different users m=
ight be assigned to different servers providing a specific service; e.g. Vo=
ice Mail Box for different users
 might be hosted by different servers, and the user need to know his assign=
ed server to be able to get access to this service. * Application Servers T=
o be able to provide service, the application server needs a list of users =
allowed to get services. The application
 server needs the following pieces of information about each user: 1. User =
ID and Credentials 2. Profile to control the level of service provided to t=
he user. If one application server provides different services using differ=
ent protocols, the application server
 might need to maintain different identities associated with these protocol=
s; e.g. an application server that provides presence service using SIP, and=
 IM service using XMPP. Ideally, an authorization framework would: 1. Off l=
oad the application server from
 the need to manage the user credentials and authenticating the user. 2. Of=
f load the application server from the need to maintain the users and their=
 profiles to control the level of service provided to the user. 3. Allow th=
e assignment of a user to a specific
 application server. * OAuth/OpenID Model With the OAuth/OpenID framework, =
the user ID and credentials will be handled by the authorization server, an=
d the application server assignment and the level of service associated wit=
h the user will be provided in the
 scope of the token associated with the user. When the application server r=
eceives a request from a user it will have all the details that would allow=
 the application server to validate the request to make sure it is coming f=
rom the right authorization server,
 and to act upon the request because it has the scope associated with the u=
ser, without the need to contact the authorization server. * Relying Party =
Regarding the Relying Party question: I think that when the user is being a=
uthenticated, that the Relying Party
 would be the SIP Client that would be given a user token (ID Token in Open=
ID lingo) to access services on behalf of the end user. So, the mapping wou=
ld be something like this: &#43;------------------------&#43;--------------=
---------------&#43; | OAuth | SIP | &#43;------------------------&#43;----=
-------------------------&#43;
 | Resource Owner | End User | | Resource Server | SIP Proxy or SIP App Ser=
ver | | Client | SIP UA (Relying Party) | | Authorization Server | Authoriz=
ation Server | &#43;------------------------&#43;--------------------------=
---&#43;</span><br>
</div>
<div><span style=3D"color: rgb(0, 0, 0); font-family: 'Courier New', Courie=
r, monospace; font-size: 14px; white-space: pre-wrap;"><br>
</span></div>
<div><br>
</div>
<div>Regards,</div>
<div>&nbsp;Rifaat</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div>&nbsp;<br>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div></div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
<div><br>
</div>
<span>
<div style=3D"font-family:Calibri;font-size:11pt;text-align:left;color:blac=
k;border-width:1pt medium medium;border-style:solid none none;padding:3pt 0=
in 0in;border-top-color:rgb(181,196,223)">
<span style=3D"font-weight:bold">From: </span>Rifaat Shekh-Yusef &lt;<a hre=
f=3D"mailto:rifaat.ietf@gmail.com" target=3D"_blank">rifaat.ietf@gmail.com<=
/a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Thursday, September 4, 2014 a=
t 8:31 AM<br>
<span style=3D"font-weight:bold">To: </span>Jon Peterson &lt;<a href=3D"mai=
lto:jon.peterson@neustar.biz" target=3D"_blank">jon.peterson@neustar.biz</a=
>&gt;<br>
<span style=3D"font-weight:bold">Cc: </span>Paul Kyzivat &lt;<a href=3D"mai=
lto:pkyzivat@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit.edu</a>&gt;,=
 &quot;<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.o=
rg</a>&quot; &lt;<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipc=
ore@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [sipcore] A SIP OAuth =
use case<br>
</div>
<div>
<div>
<div><br>
</div>
<div>
<div>
<div dir=3D"ltr"><br>
<div class=3D"gmail_extra"><br>
<br>
<div class=3D"gmail_quote">On Wed, Sep 3, 2014 at 4:00 PM, Peterson, Jon <s=
pan dir=3D"ltr">
&lt;<a href=3D"mailto:jon.peterson@neustar.biz" target=3D"_blank">jon.peter=
son@neustar.biz</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<br>
While I would echo Paul's thoughts below about the plausibility of service<=
br>
providers using this model to authorize recording, I also agree that<br>
third-party recording is an excellent example of a service architecture<br>
that requires an authorization mechanism like OAuth. An end user needs to<b=
r>
authorize a third party to participate in the session under certain<br>
constraints, and coordinate the association between the third party and<br>
the VoIP service. This example also (rightly, in my opinion) conducts the<b=
r>
majority of the flow in HTTP where it belongs.<br>
<br>
</blockquote>
<div><br>
</div>
<div>I agree with that.</div>
<div><br>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
Many of the use cases under discussion in this thread, though, are of the<b=
r>
form where there are really only two involved parties: an end user and a<br=
>
service provider. If you are my service provider, then in traditional SIP<b=
r>
I share a secret with you that I use to REGISTER my devices. I don't think<=
br>
you need OAuth for that. </blockquote>
<div><br>
</div>
<div>If you assume that there is one server that provides all the services,=
 then you are right.</div>
<div>But that is not the case all the time. I think that Dale articulated t=
hat clearly in the following response:</div>
<div><a href=3D"http://www.ietf.org/mail-archive/web/sipcore/current/msg062=
33.html" target=3D"_blank">http://www.ietf.org/mail-archive/web/sipcore/cur=
rent/msg06233.html</a><br>
</div>
<div><br>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
I do understand that OAuth 2.0 is used in the<br>
wild for SSO, though I sympathize with Matt that the base flow shown in<br>
RFC6749 Sec 1.2 just doesn't map onto the requirements for SSO.<br>
<br>
</blockquote>
<div><br>
</div>
<div>I agree that the OAuth 2.0 &quot;as is&quot; does not address the SSO =
requirements.</div>
<div>The SIP OAuth proposal is not just using OAuth 2.0 &quot;as is&quot;, =
but it is extending the framework to authenticate and provide a *user* spec=
ific token.</div>
<div><br>
</div>
<div>Since Matt mentioned OpenID few times, I looked at the OpenID specific=
ation.</div>
<div>What they are doing there is that they defined a new token, called ID =
Token, that is associated with the *user*; which is exactly what the SIP OA=
uth proposal is doing.</div>
<div><a href=3D"http://openid.net/specs/openid-connect-core-1_0.html#IDToke=
n" target=3D"_blank">http://openid.net/specs/openid-connect-core-1_0.html#I=
DToken</a><br>
</div>
<div><br>
</div>
<div>My plan is to use the same terminology, ID Token, in the next version =
of the SIP OAuth document to clarify that, and to differentiate it from the=
 access token.</div>
<div><br>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
I think, though, we might usefully break uses cases here down by who the<br=
>
intended relying party is:<br>
<br>
- The user's own SIP service provider (my enterprise, my carrier, my OTT<br=
>
provider)<br>
- A third party that a user wants to authorize for a service (recording<br>
service... But anything else?)<br>
<br>
I would also append some use cases where the relying party:<br>
<br>
- Someone with no association with the user (caller ID sorts of cases)<br>
<br>
This last is where I've argued things like IdP might be relevant.<br>
<br>
</blockquote>
<div><br>
</div>
<div>EKR has a nice description for binding IdP to OAuth in his WebRTC Secu=
rity Architecture document here:</div>
<div><a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-security-arch=
-10#appendix-A.2" target=3D"_blank">https://tools.ietf.org/html/draft-ietf-=
rtcweb-security-arch-10#appendix-A.2</a><br>
</div>
<div><br>
</div>
<div>We could use the same idea for SIP.</div>
<div><br>
</div>
<div><br>
</div>
<div>Regards,</div>
<div>&nbsp;Rifaat</div>
<div><br>
</div>
<div><br>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
Are there any other high-level buckets of use cases here?<br>
<span><font color=3D"#888888"><br>
Jon Peterson<br>
Neustar, Inc.<br>
</font></span>
<div>
<div><br>
On 8/25/14, 11:44 AM, &quot;Paul Kyzivat&quot; &lt;<a href=3D"mailto:pkyziv=
at@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit.edu</a>&gt; wrote:<br>
<br>
&gt;On 8/25/14 11:50 AM, Rifaat Shekh-Yusef wrote:<br>
&gt;&gt; Hi Matt,<br>
&gt;&gt;<br>
&gt;&gt; The use case and the solution provided assumes that the VoIP Provi=
der<br>
&gt;&gt; makes the decision on when and what to record.<br>
&gt;&gt; While this is a valid use case, there are other use cases that all=
ow the<br>
&gt;&gt; user to decide on when and what to record; take a look at RFC6341 =
for<br>
&gt;&gt; more info.<br>
&gt;&gt; <a href=3D"http://datatracker.ietf.org/doc/rfc6341/" target=3D"_bl=
ank">http://datatracker.ietf.org/doc/rfc6341/</a><br>
&gt;<br>
&gt;SIPREC supports numerous topologies.<br>
&gt;It would support the one that Matt outlines as well as end-user<br>
&gt;controlled ones.<br>
&gt;<br>
&gt;I found Matt's use case interesting. I think it would be nice if such<b=
r>
&gt;services were available in that form. But I have difficulty imagining<b=
r>
&gt;any of the VoIP providers I'm aware of supporting this model. My<br>
&gt;impression is that this means that they are giving up a value added<br>
&gt;revenue market, that they are loath to do. The most likely justificatio=
n<br>
&gt;I can imagine that they don't want to legal consequences of doing<br>
&gt;recording.<br>
&gt;<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp;Thanks,<br>
&gt;&nbsp; &nbsp; &nbsp; &nbsp;Paul<br>
&gt;<br>
&gt;&gt; For these use cases, where the user is in control, you need to awa=
y to<br>
&gt;&gt; authenticate and control which users are allowed to record and the=
 level<br>
&gt;&gt; of service provided for that specific user.<br>
&gt;&gt; This is where the solutions provided in section 3 or section 4 of =
the<br>
&gt;&gt; SIP OAuth proposal come into play.<br>
&gt;&gt; Obviously, in this case the authorization server is different from=
<br>
&gt;&gt; authorization server used to establish the trust relationship betw=
een<br>
&gt;&gt; the VoIP Provider and the CRS Provider.<br>
&gt;&gt; In this case, the authorization server must authenticate the user =
and<br>
&gt;&gt; provide his UA with an access token or a code that controls the us=
er's<br>
&gt;&gt; access to the service.<br>
&gt;&gt;<br>
&gt;&gt; Regards,<br>
&gt;&gt;&nbsp; &nbsp;Rifaat<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; On Sat, Aug 23, 2014 at 9:05 AM, Matt Ryan &lt;<a href=3D"mailto:i=
etf@mvryan.org" target=3D"_blank">ietf@mvryan.org</a><br>
&gt;&gt; &lt;mailto:<a href=3D"mailto:ietf@mvryan.org" target=3D"_blank">ie=
tf@mvryan.org</a>&gt;&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;Included is the use case I spoke of before.&nbs=
p; My apologies for the<br>
&gt;&gt;delay.<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;This is written as though it could be included =
in<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;[draft-yusef-sipcore-sip-oauth] at section 6.<b=
r>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;6. Examples<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;6.1. Example: Call Recording Service<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;This example illustrates the use of SIP OAuth b=
etween three<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;parties:&nbsp; A hosted VoIP service provider, =
a Call Recording Service<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;provider, and a phone system end-user.&nbsp; In=
 this example:<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;- The phone system end-user is a customer of bo=
th the hosted VoIP<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;provider and the Call Recording Service provide=
r<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;- The hosted VoIP service provider is a standar=
d provider of hosted<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;business-class VoIP telephone service using SIP=
<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;- The Call Recording Service provider is a dist=
inct entity from the<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;hosted VoIP provider<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;The Call Recording Service provides to customer=
s both the call<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;recording abilities and management of call reco=
rdings<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;(configuration, sharing and distribution, reten=
tion, etc.).&nbsp; This<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;service can accept and record RTP traffic, asso=
ciate all RTP streams<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;with a single call together, and store and cata=
log the recorded data<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;for future searching and retrieval.&nbsp; The s=
ervice also offers a web<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;interface where customers may view and retrieve=
 stored calls.<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;Stored calls may range from simple person-to-pe=
rson voice calls to<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;multi-party conferences with a multitude of aud=
io and video<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;streams.&nbsp; In this example, customers are b=
illed based on the amount<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;of data they store, although other models are c=
ertainly possible.<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;6.1.1. OAuth 2.0 Roles<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;In this example, each party assumes the followi=
ng OAuth 2.0 roles as<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;defined in [RFC6749] Section 1.1:<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;- The end-user assumes the role of resource own=
er<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;- The hosted VoIP service provider assumes the =
role of client<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;- The Call Recording Service provider assumes t=
he role of resource<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;server as well as the role of authorization ser=
ver<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;6.1.2. Description<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;There are two parts to the example:&nbsp; Initi=
al configuration and<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;real-time recording authorization.<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;Each section includes a simplified flow diagram=
 for the purpose of<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;describing the corresponding portion of the exa=
mple.&nbsp; For the sake<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;of brevity, certain parts of the OAuth dialog h=
ave been eliminated.<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;Implements should refer to [RFC6749] for detail=
s on OAuth.<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;6.1.2.1 Initial Configuration<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &#43;-------------&#43;&nbsp; &nbsp; &n=
bsp;&#43;---------------&#43;&nbsp; &nbsp; &nbsp;&#43;--------------&#43;<b=
r>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; |&nbsp; End-User&nbsp; &nbsp;|&nbsp; &n=
bsp; &nbsp;| VoIP Provider |&nbsp; &nbsp; &nbsp;| CRS Provider |<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; | Web Browser |&nbsp; &nbsp; &nbsp;|&nb=
sp; &nbsp; Website&nbsp; &nbsp; |&nbsp; &nbsp; &nbsp;|&nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &#43;-------------&#43;&nbsp; &nbsp; &n=
bsp;&#43;---------------&#43;&nbsp; &nbsp; &nbsp;&#43;--------------&#43;<b=
r>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |&nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;| HTTP 1&nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;| (Configure=
 CRS)&nbsp; &nbsp; |&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=
 &nbsp; &nbsp; &nbsp;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|-----------=
--------&gt;|&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=
 &nbsp; &nbsp;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |&nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;| HTTP 2&nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;| (OAuth Red=
irect&nbsp; &nbsp; |&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=
 &nbsp; &nbsp; &nbsp;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; to C=
RS Website)&nbsp; &nbsp;|&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&lt;-------=
------------|&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=
 &nbsp; &nbsp;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |&nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;| HTTP 3&nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;| (Authorize=
 SIP from VoIP provider&nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|-----------=
------------------------------&gt;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |&nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;| HTTP 4&nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;| (OAuth Red=
irect back to VoIP portal)&nbsp; &nbsp; &nbsp;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&lt;-------=
----------------------------------|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |&nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;| HTTP 5&nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|-----------=
--------&gt;|&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=
 &nbsp; &nbsp;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | HTTP 6&nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | (Request Acces=
s and |<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |&nbsp; refresh =
tokens)&nbsp; &nbsp; |<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |---------------=
-----&gt;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |&nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;Some time after having signed up for both servi=
ces, but prior to<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;attempting to record any calls, the end-user lo=
gs into the web<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;portal of the hosted VoIP provider with a web b=
rowser and configures<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;their call recording service (HTTP 1).&nbsp; Th=
is configuration includes<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;providing a URI where the call recording servic=
e may be reached,<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;along with a set of API credentials, provided b=
y the call recording<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;service, which will allow the client to initiat=
e an OAuth exchange.<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;The end-user also configures the conditions und=
er which call<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;recordings should take place.&nbsp; For example=
, the end-user may request<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;to record every multi-party (conference) call, =
or every call<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;involving a specific set of users.&nbsp; The en=
d-user may also specify<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;other restrictions, such as restricting codec n=
egotiation to G.711u<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;for audio and H.264 for video in order to recor=
d the RTP streams.<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;Once the end-user saves the configuration, the =
hosted VoIP provider<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;web portal redirects the end-user's web browser=
 to the call<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;recording service website and a standard HTTP O=
Auth dialog begins<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;(HTTP 2).&nbsp; The end-user authorizes the hos=
ted VoIP provider to<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;access (i.e. send SIP and RTP traffic to) the c=
all recording<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;service, for the purpose of recording calls, so=
 long as the<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;configured conditions are met (HTTP 3).&nbsp; T=
he result of this<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;interaction is that the hosted VoIP provider en=
ds up receiving an<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;OAuth access token and refresh token from the c=
all recording service<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;(HTTP 4, HTTP 5, HTTP 6).&nbsp; These tokens ar=
e saved in the end-user's<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;hosted VoIP account database.<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;6.1.2.2 Real-time Recording Authorization<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &#43;-------------&#43;&nbsp; &nbsp; &n=
bsp;&#43;---------------&#43;&nbsp; &nbsp; &nbsp; &#43;--------------&#43;<=
br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; |&nbsp; End-User&nbsp; &nbsp;|&nbsp; &n=
bsp; &nbsp;| VoIP Provider |&nbsp; &nbsp; &nbsp; | CRS Provider |<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; |&nbsp; SIP Phone&nbsp; |&nbsp; &nbsp; =
&nbsp;|&nbsp; &nbsp; Website&nbsp; &nbsp; |&nbsp; &nbsp; &nbsp; |&nbsp; &nb=
sp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &#43;-------------&#43;&nbsp; &nbsp; &n=
bsp;&#43;---------------&#43;&nbsp; &nbsp; &nbsp; &#43;--------------&#43;<=
br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |&nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;| SIP INVITE=
 1&nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nb=
sp; &nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|-----------=
--------&gt;|&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;=
 &nbsp; &nbsp; |<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | SIP INVITE 2&n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | (with access t=
oken)&nbsp; |<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |---------------=
------&gt;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |&nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | SIP 401 Unauth=
orized |<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |&lt;-----------=
----------|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |&nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | SIP INVITE 3&n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | (with refresh =
token) |<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |---------------=
------&gt;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |&nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | SIP 200 1&nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | (new access to=
ken)&nbsp; &nbsp;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |&lt;-----------=
----------|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |&nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | SIP INVITE 4&n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | (with access t=
oken)&nbsp; |<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |---------------=
------&gt;|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |&nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | SIP 200 2&nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |&lt;-----------=
----------|<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|&nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |&nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;Independently of and after the initial configur=
ation phase, the<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;end-user makes a telephone call using the hoste=
d VoIP provider (SIP<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;INVITE 1).&nbsp; When this call takes place, th=
e hosted VoIP provider<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;looks to see whether it believes this call shou=
ld be recorded.&nbsp; If<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;so, at this point it would branch the call sess=
ion to the call<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;recording service, using SIP OAuth to pass the =
previously provided<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;access token for authorization (SIP INVITE 2).&=
nbsp; Once the access<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;token is validated by the call recording servic=
e (perhaps after<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;first providing a new access token based on rec=
eipt of a valid<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;refresh token), the call recording service will=
 check the rights<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;that were previously authorized and examine the=
 SIP to determine<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;whether it can accept the call.&nbsp; If so, it=
 completes the<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;establishment of the session and begins receivi=
ng and recording the<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;RTP stream(s) (SIP 200 OK).&nbsp; The call reco=
rding service provider<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;could also deny the request, either because the=
 tokens are invalid<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;or because the content of<br>
&gt;&gt;&nbsp; &nbsp; &nbsp; &nbsp;the SIP invite does not match the previo=
usly authorized<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;conditions, in which case a SIP 403 would be re=
turned by the call<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;recording service provider and the call would n=
ot be recorded -<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;however, the initial call branch would continue=
 without<br>
&gt;&gt;interruption.<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;Note:<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;[RFC6749] specifies that when a client uses a r=
efresh token to<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;request a new access token, the response is HTT=
P 200 with the new<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;access token and optionally a new refresh token=
 included in the<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;payload.&nbsp; In this example, a SIP 200 respo=
nse (SIP 200 1) is sent<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;which contains the new token(s).&nbsp; However,=
 this could be confusing<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;as SIP 200 is generally viewed as the acceptanc=
e of the INVITE,<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;which is not what is happening in this case.&nb=
sp; Should SIP 200 be used<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;for consistency, or should another mechanism be=
 used?<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;6.1.3. Justification<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;6.1.3.1. Hosted VoIP Service Provider<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;In this example, the hosted VoIP service provid=
er can allow their<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;customers to enable call recording in their VoI=
P service by<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;selecting from any of a number of supported cal=
l recording service<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;providers.&nbsp; This allows the hosted VoIP se=
rvice provider to offer<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;this feature to customers without requiring tha=
t the hosted VoIP<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;service provider implement and support this fea=
ture themselves.<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;6.1.3.2. Call Recording Service Provider<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;In this example, the Call Recording Service pro=
vider can focus on<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;value and innovation in the area of recording c=
alls and managing<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;recorded calls without having to build a full-f=
eatured telephony<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;offering.<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;6.1.3.3. Customer<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;In this example, the customer has more flexibil=
ity in choosing a<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;complete solution that meets all of the custome=
r needs.&nbsp; The<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;customer does not have to settle for a substand=
ard call recording<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;service offering in order to obtain other featu=
res they seek in a<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;hosted VoIP provider, and vice versa.<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;6.1.4. Variants<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;A simple variant of this example is one wherein=
 one of the services<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;(either the VoIP service or the call recording =
service) is managed<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;directly by the end-user, but the other is not.=
&nbsp; For example, the<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;end-user may wish to make use of an external ho=
sted VoIP service<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;provider, but may have business or legal reason=
s that forbid the<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;end-user from allowing a third party to store a=
nd manage recorded<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;calls.&nbsp; The end-user could choose to set u=
p their own call recording<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;service as described in this example, and use S=
IP OAuth to<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;facilitate interaction between the hosted VoIP =
service and the<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;end-user's own call recording service.<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;6.2. Other SIP OAuth examples<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;Many other SIP-based services could leverage SI=
P OAuth to provide<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;value-added service to an end-user of a hosted =
VoIP service<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;provider.&nbsp; Some examples of these types of=
 services are listed<br>
&gt;&gt;below.<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;Voicemail service provider:&nbsp; The end-user =
configures their hosted<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;VoIP service provider to manage voicemail throu=
gh a separate<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;voicemail service provider, which chooses wheth=
er to store voicemail<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;based on existing quotas, Caller ID, etc.<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;Conferencing service provider:&nbsp; A conferen=
cing service may wish to<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;bill customers in a more granular fashion, for =
example, based on the<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;number of participants and attendees in a confe=
rence, the duration<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;of conference, whether video was also included,=
 which codecs were<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;being used, etc. instead of a flat monthly rate=
.&nbsp; SIP OAuth would<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;facilitate metered authorization for a client's=
 use of the service,<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;instead of all-or-nothing access.<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;Call center service provider:&nbsp; A third par=
ty may provide ring groups<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;or call queues for a hosted VoIP provider along=
 with detailed<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;reports and dashboards.&nbsp; The end-user uses=
 OAuth over SIP to control<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;who can log into which queues or ring groups, e=
tc.<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;--<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;Matt Ryan<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;code slinger | <a href=3D"http://zoomulus.com" =
target=3D"_blank">zoomulus.com</a> &lt;<a href=3D"http://zoomulus.com" targ=
et=3D"_blank">http://zoomulus.com</a>&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;ietf at mvryan dot org<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;_______________________________________________=
<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;sipcore mailing list<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;<a href=3D"mailto:sipcore@ietf.org" target=3D"_=
blank">sipcore@ietf.org</a> &lt;mailto:<a href=3D"mailto:sipcore@ietf.org" =
target=3D"_blank">sipcore@ietf.org</a>&gt;<br>
&gt;&gt;&nbsp; &nbsp; &nbsp;<a href=3D"https://www.ietf.org/mailman/listinf=
o/sipcore" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sipcore<=
/a><br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; _______________________________________________<br>
&gt;&gt; sipcore mailing list<br>
&gt;&gt; <a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf=
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&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=
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&gt;sipcore mailing list<br>
&gt;<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org<=
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&gt;<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/sipcore</a><br>
<br>
_______________________________________________<br>
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<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a><=
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>https://www.ietf.org/mailman/listinfo/sipcore</a><br>
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</blockquote>
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</div>
</div>
</div>
</span></div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</div>
</div>
</div>
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</blockquote>
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--_000_D059A86A134723jonpetersonneustarbiz_--


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Subject: Re: [sipcore] A SIP OAuth use case
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Thanks for the detailed response Jon.
Please, see my reply inline...

Regards,
 Rifaat


On Tue, Oct 7, 2014 at 7:01 PM, Peterson, Jon <jon.peterson@neustar.biz>
wrote:

>
>  I know I've owed you a response to this one for a while... Still just
> trying to understand the requirements.
>
>  I suspect that application services in the enterprise, the ones that you
> worry about maintaining ACLs, don't get to avoid having "a separate user
> directory" so easily. A classic voicemail service, for example, needs to
> keep voicemails for Alice separate from Bob - that's why typically it
> authenticates who Alice and Bob are.
>

I obviously agree that the service must keep separate boxes for their users
and it must authenticate the user to allow access to a specific box, but I
am not sure that the authentication must be done by the voicemail server
itself.
The authentication could be done by an Authorization Server and as a result
the SIP endpoint would be given an ID Token, as per OpenID, which proves
the identity of the end user. The SIP endpoint will be able to use the ID
Token to get access to the user's box. The voicemail server will be able to
identify the user and verify the validity of the ID Token because the token
has the signature of an Authorization Server that the voicemail server
trusts.



> Such a service must have effectively a "directory" of separate users that
> effectively implies ACLs.
>
>
The ID Token could also have a scope that controls the services and level
of service provided to the user.




>  Now if, as you say:
>
>  > The application server needs the following pieces of information about
> each user:
> > 1. ID and credentials
> > 2. User profile to control the level of service provided to the user.
>
>  What is your app server using that ID and credentials for, if not to
> compare to something that is effectively an ACL - like, to prevent Alice
> from getting Bob's voicemail? My point earlier was that if all you care
> about is showing that someone is Alice or Bob (or just that they're an
> authenticated user and it doesn't matter who), then a solution to the
> requirements could look very similar to RFC4474 or STIR: you just need
> authentication, not authorization, and calling it authorization is just
> confusing the issue.
>
>
I care about both, authentication and authorization, and my argument is
that an ID Token could be used to provide the needed user authentication,
and the scope associated with the ID Token could be used to provide the
authorization to control access to services for that specific user.



>  But, there's a different approach here, one where a service doesn't need
> to know IDs or credentials or anything about users.
>

In this specific case, the service obviously needs to know the user ID, but
it does not have to know the user's credentials to authorize access to the
voicemail box.



> This is an approach where I just get a token from an authorization
> service, and hand that token to the app service, and the app service uses
> that token *instead of* my ID or credentials or any personal profile about
> me.
>

As I mentioned above, the ID Token proves your identity and has a scope
that controls your access to services.



> I can imagine a very odd voicemail service that, rather than putting
> messages into buckets for Alice and Bob, put them all into one generic
> bucket - and then an outside authorization service kept track of the fact
> that messages 2, 17, and 109 are for Alice. When Alice requests access to
> her voicemail from the authorization service, it gives her a token that
> proves to the voicemail service that she is allowed to access only messages
> 2, 17, and 109, and she can only do it in the next five minutes, say, to
> prevent capture and replay. The voicemail service then has no idea who she
> is or who those messages are for, it just ponies them up. If *that* is what
> you said above, at least I would understand why you want to use OAuth - it
> would be a terrible architecture (the authZ service keeps track of those
> messages how exactly?), but at least this is what OAuth is for. The
> architectures make more sense with third parties.
>
>
That obviously does not make sense for a voicemail service,

But, let's take a look at a different example; an adhoc conference server:
I can imagine an adhoc conference server that accepts requests to allocate
a focus to a user based on the ID Token provided in the incoming request.
The ID Token proves the identity of the user and provides a scope to
control the level of service provided to the user. The conference server
will be able to verify the validity of the ID Token because the token has
the signature of an Authorization Server that the conference server trusts.
In this specific case, do you see a need for the conference server to
maintain a user directory?

In any case, I am not trying to completely get rid of user directory from
all services, but mainly to get rid of the need to maintain separate user
identities and credentials per service.
Ideally, I would like the user to have one identity and be able to use that
identity to authenticate and get access to all services in the enterprise.


 A third approach would be problems we looked at as "trait-based"
> authorization problems. The classic example is where for some service,
> perhaps it matters whether Alice is in the Engineering department or the
> Finance department, rather than who Alice is exactly. Maybe those're the
> sort of elements you mean by "user profile." We did a whole requirements
> process years ago on "trait-based" authorization mechanisms intended to
> satisfy cases like this: see RFC4484. You'll note it shows architectures
> where a SIP endpoints point by-reference to HTTP objects that contain
> authorization policy elements like such "traits." There are a bunch of
> solutions for that space, too, but if that's what you want, then my
> question for you would be, do you have any requirements beyond what's
> defined in RFC4484 Sec. 5?
>
>
I have never read RFC4484 before; I will take a look.

Regards,
 Rifaat





>  Jon Peterson
> Neustar, Inc.
>
>   From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
> Date: Wednesday, September 17, 2014 at 6:22 AM
>
> To: Jon Peterson <jon.peterson@neustar.biz>
> Cc: Paul Kyzivat <pkyzivat@alum.mit.edu>, "sipcore@ietf.org" <
> sipcore@ietf.org>
> Subject: Re: [sipcore] A SIP OAuth use case
>
>
>
> On Mon, Sep 15, 2014 at 3:29 PM, Peterson, Jon <jon.peterson@neustar.biz>
> wrote:
>
>>
>>  Format snags aside,
>>
>
>  I copied that text from a different editor.
> The format seems ok here:
> http://www.ietf.org/mail-archive/web/sipcore/current/msg06258.html
>
>  So, it might be something on your side?
>
>
>
>>  I have a bit of difficulty even differentiating the issues listed
>> below: surely the reason why enterprises assign multiple identities per
>> user is to authorize access to different services separately.
>>
>
>  Sorry, I do not get that.
> Why would the enterprise be required to create multiple identities to
> allow access to different services separately? in other words, why cannot
> the enterprise create one identity and still be able to control the user's
> access to different services provided by different servers?
>
>
>
>>  This is an issue that could be resolved with an access control list per
>> service and a trusted identity broker in the enterprise vouching for user
>> identities to these services.
>>
>
>  But that means that each server that provides a service must maintain a
> separate user directory, which is exactly what I am trying to avoid.
>
>
>
>>  But to convince ourselves that one approach to this is better than
>> another, we need to look at a level of detail below these issue
>> descriptions. Again, in order to access a voicemail service in an
>> enterprise, say, what does the voicemail service need to know about the
>> user?
>>
>>  Like, for regular access to the voicemail for "jon@enterprise," surely
>> all the voicemail service needs to know is that the entity authoritative
>> for the @enterprise issued me the name "jon." Maybe the "admin" account
>> gets to access all the voicemails, if need be.
>>
>
>  The application server needs the following pieces of information about
> each user:
> 1. ID and credentials
> 2. User profile to control the level of service provided to the user.
>
>  You cannot assume that all users will get the same level of service.
>
>
>
>>  There are a variety of mechanisms that suffice to prove simple
>> identities like that to a service.
>>
>
>  Can you elaborate?
> Are you assuming that the service still have a separate user directory?
>
>
>
>>  If, however, I as "jon" needed for a third-party transcription service
>> to access a subset of the messages in my voicemail box, and I wanted the
>> permissions for that access to expire as soon as the messages had been
>> transcribed, then there is a much narrower set of mechanisms that could
>> provide that capability.
>>
>
>  You keep going back to the OAuth 2.0 model and try to impose it on SIP,
> which would work in some use cases.
>
>  What I am proposing is something along the lines of OpenID, which
> extends the OAuth to authenticate the *user*.
> In this case, the mapping between OAuth and SIP would be something like
> the following:
>
>  OAuth                                    SIP
> ---------------------------------------------------------------------------
> Resource Owner                     End User
> Resource Server                     SIP Proxy or SIP Application Server
> Client                                     SIP UA
> Authorization Server                Authorization Server
>
>  Do you see a problem with this model?
>
>
>
>>  But it sounds to me, from this issues list, like your requirements are
>> more along the former lines than the latter.
>>
>>
>  Actually, it is both.
>
>  Here is an example of a use case that fits with the OAuth 2.0 model:
>
>  An Application Server that provides Mobile Clients with access to SIP
> services.
> The Mobile Client uses a proprietary protocol to communicate with the
> Application Server that registers and gets service from the SIP network on
> behalf of the user that is using the Mobile Client.
>
>  The communication channels between the players in this case are as
> follows (hopefully the format is ok this time):
>
>  [Mobile Client]  <----------->  [Application Server] <-----------> [SIP
> Proxy]
>           /\                                         /\
>            |                                          |
>            |                                         \/
>             ------------------------> [Authorization Server]
>
>
>  In this case, the mapping between OAuth and SIP would be something like
> the following:
>
>  OAuth                                    SIP
> ---------------------------------------------------------------------------
> Resource Owner                     End User
> Resource Server                     SIP Proxy
> Client                                     Application Server (providing
> service to the mobile client)
> Authorization Server                Authorization Server
>
>
>  Regards,
>  Rifaat
>
>
>
>
>
>>  Jon Peterson
>> Neustar, Inc.
>>
>>   From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
>> Date: Saturday, September 13, 2014 at 12:32 PM
>>
>> To: Jon Peterson <jon.peterson@neustar.biz>
>> Cc: Paul Kyzivat <pkyzivat@alum.mit.edu>, "sipcore@ietf.org" <
>> sipcore@ietf.org>
>> Subject: Re: [sipcore] A SIP OAuth use case
>>
>>
>>
>> On Thu, Sep 4, 2014 at 7:27 PM, Peterson, Jon <jon.peterson@neustar.biz>
>> wrote:
>>
>>>
>>>  I understand that OAuth can be altered from its basic flows to provide
>>> tokens for OpenID, and IdP could be bound to OAuth, and while this is all
>>> very interesting it doesn't change much about the requirements discussion I
>>> think we need to have: the one about what kinds of use cases we want to
>>> satisfy and who the relying parties are in those cases. Once we understand
>>> that, we will understand what tools are applicable to these use cases -
>>> preferably, not tools that need to be twisted into a different shape to be
>>> applicable.
>>>
>>>
>>  Well, OpenID extends the OAuth 2.0 Framework to authenticate the user;
>> why do you think it is "twisting" OAuth into a different shape?
>>
>>
>>
>>
>>>  It would helpful, in the enterprise SSO case, for us to understand
>>> what kinds of application servers a user might need to access, and what
>>> exactly those application servers need to know about the user in order to
>>> accomplish their function.
>>>
>>>
>>  Below is a description for some enterprise issues, that hopefully will
>> be addressed by a new authorization framework:
>>
>>
>>  * Multiple Identities One of the major issues in the enterprise is the
>> need for a user to obtain multiple uncorrelated identities associated with
>> different products to get access to the various services provided by the
>> enterprise. For example, in the enterprise the user will be provided with
>> different identities to: o Login his device (PC, laptop, mobile, ...) to
>> the network. o Register to the SIP network and get basic SIP services. o
>> Access his Voice Mail. o Host a conference. o etc Typically, these
>> identities belong to different administrative domains and realms. Instead
>> of these multiple identities, ideally the enterprise would like to create
>> and maintain only one identity for the user and then allow the user access
>> to all SIP and non-SIP services using that one identity. * Authorization of
>> Access to Services Another issue in the enterprise is the need to control
>> and authorize a user access to services. Because application servers have
>> their own user directory, the control of access to services is spread over
>> a multitude of servers in the network. Ideally, the enterprise would like
>> to manage the access to services using one centralized entity and for the
>> various entities that provide the service to become the enforcement points.
>> * Assignment of Services In the enterprise, different users might be
>> assigned to different servers providing a specific service; e.g. Voice Mail
>> Box for different users might be hosted by different servers, and the user
>> need to know his assigned server to be able to get access to this service.
>> * Application Servers To be able to provide service, the application server
>> needs a list of users allowed to get services. The application server needs
>> the following pieces of information about each user: 1. User ID and
>> Credentials 2. Profile to control the level of service provided to the
>> user. If one application server provides different services using different
>> protocols, the application server might need to maintain different
>> identities associated with these protocols; e.g. an application server that
>> provides presence service using SIP, and IM service using XMPP. Ideally, an
>> authorization framework would: 1. Off load the application server from the
>> need to manage the user credentials and authenticating the user. 2. Off
>> load the application server from the need to maintain the users and their
>> profiles to control the level of service provided to the user. 3. Allow the
>> assignment of a user to a specific application server. * OAuth/OpenID Model
>> With the OAuth/OpenID framework, the user ID and credentials will be
>> handled by the authorization server, and the application server assignment
>> and the level of service associated with the user will be provided in the
>> scope of the token associated with the user. When the application server
>> receives a request from a user it will have all the details that would
>> allow the application server to validate the request to make sure it is
>> coming from the right authorization server, and to act upon the request
>> because it has the scope associated with the user, without the need to
>> contact the authorization server. * Relying Party Regarding the Relying
>> Party question: I think that when the user is being authenticated, that the
>> Relying Party would be the SIP Client that would be given a user token (ID
>> Token in OpenID lingo) to access services on behalf of the end user. So,
>> the mapping would be something like this:
>> +------------------------+-----------------------------+ | OAuth | SIP |
>> +------------------------+-----------------------------+ | Resource Owner |
>> End User | | Resource Server | SIP Proxy or SIP App Server | | Client | SIP
>> UA (Relying Party) | | Authorization Server | Authorization Server |
>> +------------------------+-----------------------------+
>>
>>
>>  Regards,
>>  Rifaat
>>
>>
>>
>>
>>
>>
>>
>>>  Jon Peterson
>>> Neustar, Inc.
>>>
>>>   From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
>>> Date: Thursday, September 4, 2014 at 8:31 AM
>>> To: Jon Peterson <jon.peterson@neustar.biz>
>>> Cc: Paul Kyzivat <pkyzivat@alum.mit.edu>, "sipcore@ietf.org" <
>>> sipcore@ietf.org>
>>> Subject: Re: [sipcore] A SIP OAuth use case
>>>
>>>
>>>
>>>
>>> On Wed, Sep 3, 2014 at 4:00 PM, Peterson, Jon <jon.peterson@neustar.biz>
>>> wrote:
>>>
>>>>
>>>> While I would echo Paul's thoughts below about the plausibility of
>>>> service
>>>> providers using this model to authorize recording, I also agree that
>>>> third-party recording is an excellent example of a service architecture
>>>> that requires an authorization mechanism like OAuth. An end user needs
>>>> to
>>>> authorize a third party to participate in the session under certain
>>>> constraints, and coordinate the association between the third party and
>>>> the VoIP service. This example also (rightly, in my opinion) conducts
>>>> the
>>>> majority of the flow in HTTP where it belongs.
>>>>
>>>>
>>>  I agree with that.
>>>
>>>
>>>
>>>> Many of the use cases under discussion in this thread, though, are of
>>>> the
>>>> form where there are really only two involved parties: an end user and a
>>>> service provider. If you are my service provider, then in traditional
>>>> SIP
>>>> I share a secret with you that I use to REGISTER my devices. I don't
>>>> think
>>>> you need OAuth for that.
>>>
>>>
>>>  If you assume that there is one server that provides all the services,
>>> then you are right.
>>> But that is not the case all the time. I think that Dale articulated
>>> that clearly in the following response:
>>> http://www.ietf.org/mail-archive/web/sipcore/current/msg06233.html
>>>
>>>
>>>
>>>> I do understand that OAuth 2.0 is used in the
>>>> wild for SSO, though I sympathize with Matt that the base flow shown in
>>>> RFC6749 Sec 1.2 just doesn't map onto the requirements for SSO.
>>>>
>>>>
>>>  I agree that the OAuth 2.0 "as is" does not address the SSO
>>> requirements.
>>> The SIP OAuth proposal is not just using OAuth 2.0 "as is", but it is
>>> extending the framework to authenticate and provide a *user* specific token.
>>>
>>>  Since Matt mentioned OpenID few times, I looked at the OpenID
>>> specification.
>>> What they are doing there is that they defined a new token, called ID
>>> Token, that is associated with the *user*; which is exactly what the SIP
>>> OAuth proposal is doing.
>>> http://openid.net/specs/openid-connect-core-1_0.html#IDToken
>>>
>>>  My plan is to use the same terminology, ID Token, in the next version
>>> of the SIP OAuth document to clarify that, and to differentiate it from the
>>> access token.
>>>
>>>
>>>
>>>> I think, though, we might usefully break uses cases here down by who the
>>>> intended relying party is:
>>>>
>>>> - The user's own SIP service provider (my enterprise, my carrier, my OTT
>>>> provider)
>>>> - A third party that a user wants to authorize for a service (recording
>>>> service... But anything else?)
>>>>
>>>> I would also append some use cases where the relying party:
>>>>
>>>> - Someone with no association with the user (caller ID sorts of cases)
>>>>
>>>> This last is where I've argued things like IdP might be relevant.
>>>>
>>>>
>>>  EKR has a nice description for binding IdP to OAuth in his WebRTC
>>> Security Architecture document here:
>>>
>>> https://tools.ietf.org/html/draft-ietf-rtcweb-security-arch-10#appendix-A.2
>>>
>>>  We could use the same idea for SIP.
>>>
>>>
>>>  Regards,
>>>  Rifaat
>>>
>>>
>>>
>>>
>>>> Are there any other high-level buckets of use cases here?
>>>>
>>>> Jon Peterson
>>>> Neustar, Inc.
>>>>
>>>> On 8/25/14, 11:44 AM, "Paul Kyzivat" <pkyzivat@alum.mit.edu> wrote:
>>>>
>>>> >On 8/25/14 11:50 AM, Rifaat Shekh-Yusef wrote:
>>>> >> Hi Matt,
>>>> >>
>>>> >> The use case and the solution provided assumes that the VoIP Provider
>>>> >> makes the decision on when and what to record.
>>>> >> While this is a valid use case, there are other use cases that allow
>>>> the
>>>> >> user to decide on when and what to record; take a look at RFC6341 for
>>>> >> more info.
>>>> >> http://datatracker.ietf.org/doc/rfc6341/
>>>> >
>>>> >SIPREC supports numerous topologies.
>>>> >It would support the one that Matt outlines as well as end-user
>>>> >controlled ones.
>>>> >
>>>> >I found Matt's use case interesting. I think it would be nice if such
>>>> >services were available in that form. But I have difficulty imagining
>>>> >any of the VoIP providers I'm aware of supporting this model. My
>>>> >impression is that this means that they are giving up a value added
>>>> >revenue market, that they are loath to do. The most likely
>>>> justification
>>>> >I can imagine that they don't want to legal consequences of doing
>>>> >recording.
>>>> >
>>>> >       Thanks,
>>>> >       Paul
>>>> >
>>>> >> For these use cases, where the user is in control, you need to away
>>>> to
>>>> >> authenticate and control which users are allowed to record and the
>>>> level
>>>> >> of service provided for that specific user.
>>>> >> This is where the solutions provided in section 3 or section 4 of the
>>>> >> SIP OAuth proposal come into play.
>>>> >> Obviously, in this case the authorization server is different from
>>>> >> authorization server used to establish the trust relationship between
>>>> >> the VoIP Provider and the CRS Provider.
>>>> >> In this case, the authorization server must authenticate the user and
>>>> >> provide his UA with an access token or a code that controls the
>>>> user's
>>>> >> access to the service.
>>>> >>
>>>> >> Regards,
>>>> >>   Rifaat
>>>> >>
>>>> >>
>>>> >>
>>>> >> On Sat, Aug 23, 2014 at 9:05 AM, Matt Ryan <ietf@mvryan.org
>>>> >> <mailto:ietf@mvryan.org>> wrote:
>>>> >>
>>>> >>     Included is the use case I spoke of before.  My apologies for the
>>>> >>delay.
>>>> >>
>>>> >>     This is written as though it could be included in
>>>> >>     [draft-yusef-sipcore-sip-oauth] at section 6.
>>>> >>
>>>> >>
>>>> >>     6. Examples
>>>> >>
>>>> >>     6.1. Example: Call Recording Service
>>>> >>
>>>> >>     This example illustrates the use of SIP OAuth between three
>>>> >>     parties:  A hosted VoIP service provider, a Call Recording
>>>> Service
>>>> >>     provider, and a phone system end-user.  In this example:
>>>> >>     - The phone system end-user is a customer of both the hosted VoIP
>>>> >>     provider and the Call Recording Service provider
>>>> >>     - The hosted VoIP service provider is a standard provider of
>>>> hosted
>>>> >>     business-class VoIP telephone service using SIP
>>>> >>     - The Call Recording Service provider is a distinct entity from
>>>> the
>>>> >>     hosted VoIP provider
>>>> >>
>>>> >>     The Call Recording Service provides to customers both the call
>>>> >>     recording abilities and management of call recordings
>>>> >>     (configuration, sharing and distribution, retention, etc.).  This
>>>> >>     service can accept and record RTP traffic, associate all RTP
>>>> streams
>>>> >>     with a single call together, and store and catalog the recorded
>>>> data
>>>> >>     for future searching and retrieval.  The service also offers a
>>>> web
>>>> >>     interface where customers may view and retrieve stored calls.
>>>> >>     Stored calls may range from simple person-to-person voice calls
>>>> to
>>>> >>     multi-party conferences with a multitude of audio and video
>>>> >>     streams.  In this example, customers are billed based on the
>>>> amount
>>>> >>     of data they store, although other models are certainly possible.
>>>> >>
>>>> >>     6.1.1. OAuth 2.0 Roles
>>>> >>
>>>> >>     In this example, each party assumes the following OAuth 2.0
>>>> roles as
>>>> >>     defined in [RFC6749] Section 1.1:
>>>> >>     - The end-user assumes the role of resource owner
>>>> >>     - The hosted VoIP service provider assumes the role of client
>>>> >>     - The Call Recording Service provider assumes the role of
>>>> resource
>>>> >>     server as well as the role of authorization server
>>>> >>
>>>> >>     6.1.2. Description
>>>> >>
>>>> >>     There are two parts to the example:  Initial configuration and
>>>> >>     real-time recording authorization.
>>>> >>
>>>> >>     Each section includes a simplified flow diagram for the purpose
>>>> of
>>>> >>     describing the corresponding portion of the example.  For the
>>>> sake
>>>> >>     of brevity, certain parts of the OAuth dialog have been
>>>> eliminated.
>>>> >>     Implements should refer to [RFC6749] for details on OAuth.
>>>> >>
>>>> >>     6.1.2.1 Initial Configuration
>>>> >>
>>>> >>        +-------------+     +---------------+     +--------------+
>>>> >>        |  End-User   |     | VoIP Provider |     | CRS Provider |
>>>> >>        | Web Browser |     |    Website    |     |              |
>>>> >>        +-------------+     +---------------+     +--------------+
>>>> >>               |                    |                     |
>>>> >>               | HTTP 1             |                     |
>>>> >>               | (Configure CRS)    |                     |
>>>> >>               |------------------->|                     |
>>>> >>               |                    |                     |
>>>> >>               | HTTP 2             |                     |
>>>> >>               | (OAuth Redirect    |                     |
>>>> >>               |  to CRS Website)   |                     |
>>>> >>               |<-------------------|                     |
>>>> >>               |                    |                     |
>>>> >>               | HTTP 3             |                     |
>>>> >>               | (Authorize SIP from VoIP provider        |
>>>> >>               |----------------------------------------->|
>>>> >>               |                    |                     |
>>>> >>               | HTTP 4             |                     |
>>>> >>               | (OAuth Redirect back to VoIP portal)     |
>>>> >>               |<-----------------------------------------|
>>>> >>               |                    |                     |
>>>> >>               | HTTP 5             |                     |
>>>> >>               |------------------->|                     |
>>>> >>               |                    | HTTP 6              |
>>>> >>               |                    | (Request Access and |
>>>> >>               |                    |  refresh tokens)    |
>>>> >>               |                    |-------------------->|
>>>> >>               |                    |                     |
>>>> >>
>>>> >>     Some time after having signed up for both services, but prior to
>>>> >>     attempting to record any calls, the end-user logs into the web
>>>> >>     portal of the hosted VoIP provider with a web browser and
>>>> configures
>>>> >>     their call recording service (HTTP 1).  This configuration
>>>> includes
>>>> >>     providing a URI where the call recording service may be reached,
>>>> >>     along with a set of API credentials, provided by the call
>>>> recording
>>>> >>     service, which will allow the client to initiate an OAuth
>>>> exchange.
>>>> >>     The end-user also configures the conditions under which call
>>>> >>     recordings should take place.  For example, the end-user may
>>>> request
>>>> >>     to record every multi-party (conference) call, or every call
>>>> >>     involving a specific set of users.  The end-user may also specify
>>>> >>     other restrictions, such as restricting codec negotiation to
>>>> G.711u
>>>> >>     for audio and H.264 for video in order to record the RTP streams.
>>>> >>     Once the end-user saves the configuration, the hosted VoIP
>>>> provider
>>>> >>     web portal redirects the end-user's web browser to the call
>>>> >>     recording service website and a standard HTTP OAuth dialog begins
>>>> >>     (HTTP 2).  The end-user authorizes the hosted VoIP provider to
>>>> >>     access (i.e. send SIP and RTP traffic to) the call recording
>>>> >>     service, for the purpose of recording calls, so long as the
>>>> >>     configured conditions are met (HTTP 3).  The result of this
>>>> >>     interaction is that the hosted VoIP provider ends up receiving an
>>>> >>     OAuth access token and refresh token from the call recording
>>>> service
>>>> >>     (HTTP 4, HTTP 5, HTTP 6).  These tokens are saved in the
>>>> end-user's
>>>> >>     hosted VoIP account database.
>>>> >>
>>>> >>     6.1.2.2 Real-time Recording Authorization
>>>> >>
>>>> >>        +-------------+     +---------------+      +--------------+
>>>> >>        |  End-User   |     | VoIP Provider |      | CRS Provider |
>>>> >>        |  SIP Phone  |     |    Website    |      |              |
>>>> >>        +-------------+     +---------------+      +--------------+
>>>> >>               |                    |                      |
>>>> >>               | SIP INVITE 1       |                      |
>>>> >>               |------------------->|                      |
>>>> >>               |                    | SIP INVITE 2         |
>>>> >>               |                    | (with access token)  |
>>>> >>               |                    |--------------------->|
>>>> >>               |                    |                      |
>>>> >>               |                    | SIP 401 Unauthorized |
>>>> >>               |                    |<---------------------|
>>>> >>               |                    |                      |
>>>> >>               |                    | SIP INVITE 3         |
>>>> >>               |                    | (with refresh token) |
>>>> >>               |                    |--------------------->|
>>>> >>               |                    |                      |
>>>> >>               |                    | SIP 200 1            |
>>>> >>               |                    | (new access token)   |
>>>> >>               |                    |<---------------------|
>>>> >>               |                    |                      |
>>>> >>               |                    | SIP INVITE 4         |
>>>> >>               |                    | (with access token)  |
>>>> >>               |                    |--------------------->|
>>>> >>               |                    |                      |
>>>> >>               |                    | SIP 200 2            |
>>>> >>               |                    |<---------------------|
>>>> >>               |                    |                      |
>>>> >>
>>>> >>     Independently of and after the initial configuration phase, the
>>>> >>     end-user makes a telephone call using the hosted VoIP provider
>>>> (SIP
>>>> >>     INVITE 1).  When this call takes place, the hosted VoIP provider
>>>> >>     looks to see whether it believes this call should be recorded.
>>>> If
>>>> >>     so, at this point it would branch the call session to the call
>>>> >>     recording service, using SIP OAuth to pass the previously
>>>> provided
>>>> >>     access token for authorization (SIP INVITE 2).  Once the access
>>>> >>     token is validated by the call recording service (perhaps after
>>>> >>     first providing a new access token based on receipt of a valid
>>>> >>     refresh token), the call recording service will check the rights
>>>> >>     that were previously authorized and examine the SIP to determine
>>>> >>     whether it can accept the call.  If so, it completes the
>>>> >>     establishment of the session and begins receiving and recording
>>>> the
>>>> >>     RTP stream(s) (SIP 200 OK).  The call recording service provider
>>>> >>     could also deny the request, either because the tokens are
>>>> invalid
>>>> >>     or because the content of
>>>> >>       the SIP invite does not match the previously authorized
>>>> >>     conditions, in which case a SIP 403 would be returned by the call
>>>> >>     recording service provider and the call would not be recorded -
>>>> >>     however, the initial call branch would continue without
>>>> >>interruption.
>>>> >>
>>>> >>     Note:
>>>> >>     [RFC6749] specifies that when a client uses a refresh token to
>>>> >>     request a new access token, the response is HTTP 200 with the new
>>>> >>     access token and optionally a new refresh token included in the
>>>> >>     payload.  In this example, a SIP 200 response (SIP 200 1) is sent
>>>> >>     which contains the new token(s).  However, this could be
>>>> confusing
>>>> >>     as SIP 200 is generally viewed as the acceptance of the INVITE,
>>>> >>     which is not what is happening in this case.  Should SIP 200 be
>>>> used
>>>> >>     for consistency, or should another mechanism be used?
>>>> >>
>>>> >>     6.1.3. Justification
>>>> >>
>>>> >>     6.1.3.1. Hosted VoIP Service Provider
>>>> >>
>>>> >>     In this example, the hosted VoIP service provider can allow their
>>>> >>     customers to enable call recording in their VoIP service by
>>>> >>     selecting from any of a number of supported call recording
>>>> service
>>>> >>     providers.  This allows the hosted VoIP service provider to offer
>>>> >>     this feature to customers without requiring that the hosted VoIP
>>>> >>     service provider implement and support this feature themselves.
>>>> >>
>>>> >>     6.1.3.2. Call Recording Service Provider
>>>> >>
>>>> >>     In this example, the Call Recording Service provider can focus on
>>>> >>     value and innovation in the area of recording calls and managing
>>>> >>     recorded calls without having to build a full-featured telephony
>>>> >>     offering.
>>>> >>
>>>> >>     6.1.3.3. Customer
>>>> >>
>>>> >>     In this example, the customer has more flexibility in choosing a
>>>> >>     complete solution that meets all of the customer needs.  The
>>>> >>     customer does not have to settle for a substandard call recording
>>>> >>     service offering in order to obtain other features they seek in a
>>>> >>     hosted VoIP provider, and vice versa.
>>>> >>
>>>> >>     6.1.4. Variants
>>>> >>
>>>> >>     A simple variant of this example is one wherein one of the
>>>> services
>>>> >>     (either the VoIP service or the call recording service) is
>>>> managed
>>>> >>     directly by the end-user, but the other is not.  For example, the
>>>> >>     end-user may wish to make use of an external hosted VoIP service
>>>> >>     provider, but may have business or legal reasons that forbid the
>>>> >>     end-user from allowing a third party to store and manage recorded
>>>> >>     calls.  The end-user could choose to set up their own call
>>>> recording
>>>> >>     service as described in this example, and use SIP OAuth to
>>>> >>     facilitate interaction between the hosted VoIP service and the
>>>> >>     end-user's own call recording service.
>>>> >>
>>>> >>     6.2. Other SIP OAuth examples
>>>> >>
>>>> >>     Many other SIP-based services could leverage SIP OAuth to provide
>>>> >>     value-added service to an end-user of a hosted VoIP service
>>>> >>     provider.  Some examples of these types of services are listed
>>>> >>below.
>>>> >>
>>>> >>     Voicemail service provider:  The end-user configures their hosted
>>>> >>     VoIP service provider to manage voicemail through a separate
>>>> >>     voicemail service provider, which chooses whether to store
>>>> voicemail
>>>> >>     based on existing quotas, Caller ID, etc.
>>>> >>
>>>> >>     Conferencing service provider:  A conferencing service may wish
>>>> to
>>>> >>     bill customers in a more granular fashion, for example, based on
>>>> the
>>>> >>     number of participants and attendees in a conference, the
>>>> duration
>>>> >>     of conference, whether video was also included, which codecs were
>>>> >>     being used, etc. instead of a flat monthly rate.  SIP OAuth would
>>>> >>     facilitate metered authorization for a client's use of the
>>>> service,
>>>> >>     instead of all-or-nothing access.
>>>> >>
>>>> >>     Call center service provider:  A third party may provide ring
>>>> groups
>>>> >>     or call queues for a hosted VoIP provider along with detailed
>>>> >>     reports and dashboards.  The end-user uses OAuth over SIP to
>>>> control
>>>> >>     who can log into which queues or ring groups, etc.
>>>> >>
>>>> >>
>>>> >>
>>>> >>     --
>>>> >>     Matt Ryan
>>>> >>     code slinger | zoomulus.com <http://zoomulus.com>
>>>> >>     ietf at mvryan dot org
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> >>     _______________________________________________
>>>> >>     sipcore mailing list
>>>> >>     sipcore@ietf.org <mailto:sipcore@ietf.org>
>>>> >>     https://www.ietf.org/mailman/listinfo/sipcore
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> >> _______________________________________________
>>>> >> sipcore mailing list
>>>> >> sipcore@ietf.org
>>>> >> https://www.ietf.org/mailman/listinfo/sipcore
>>>> >>
>>>> >
>>>> >_______________________________________________
>>>> >sipcore mailing list
>>>> >sipcore@ietf.org
>>>> >https://www.ietf.org/mailman/listinfo/sipcore
>>>>
>>>> _______________________________________________
>>>> sipcore mailing list
>>>> sipcore@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>>
>>>
>>>
>>
>

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<div dir=3D"ltr">Thanks for the detailed response Jon.<div>Please, see my r=
eply inline...</div><div><br></div><div>Regards,</div><div>=A0Rifaat</div><=
div><br></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On =
Tue, Oct 7, 2014 at 7:01 PM, Peterson, Jon <span dir=3D"ltr">&lt;<a href=3D=
"mailto:jon.peterson@neustar.biz" target=3D"_blank">jon.peterson@neustar.bi=
z</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margi=
n:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204=
);border-left-style:solid;padding-left:1ex">



<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div><br>
</div>
<div>I know I&#39;ve owed you a response to this one for a while... Still j=
ust trying to understand the requirements.</div>
<div><br>
</div>
<div>I suspect that application services in the enterprise, the ones that y=
ou worry about maintaining ACLs, don&#39;t get to avoid having &quot;a sepa=
rate user directory&quot; so easily. A classic voicemail service, for examp=
le, needs to keep voicemails for Alice separate
 from Bob - that&#39;s why typically it authenticates who Alice and Bob are=
. </div></div></blockquote><div><br></div><div>I obviously agree that the s=
ervice must keep separate boxes for their users and it must authenticate th=
e user to allow access to a specific box, but I am not sure that the authen=
tication must be done by the voicemail server itself.=A0</div><div>The auth=
entication could be done by an Authorization Server and as a result the SIP=
 endpoint would be given an ID Token, as per OpenID, which proves the ident=
ity of the end user. The SIP endpoint will be able to use the ID Token to g=
et access to the user&#39;s box. The voicemail server will be able to ident=
ify the user and verify the validity of the ID Token because the token has =
the signature of an Authorization Server that the voicemail server trusts.<=
/div><div><br></div><div>=A0</div><blockquote class=3D"gmail_quote" style=
=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(20=
4,204,204);border-left-style:solid;padding-left:1ex"><div style=3D"word-wra=
p:break-word;color:rgb(0,0,0);font-size:14px;font-family:Calibri,sans-serif=
"><div>Such a service must have effectively a &quot;directory&quot; of sepa=
rate users that effectively implies ACLs.</div>
<div><br></div></div></blockquote><div><br></div><div>The ID Token could al=
so have a scope that controls the services and level of service provided to=
 the user.</div><div><br></div><div><br></div><div>=A0</div><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;=
border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex=
"><div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-f=
amily:Calibri,sans-serif"><div>
</div>
<div>Now if, as you say:</div><span>
<div><br>
</div>
<div>
<div>&gt; The application server needs the following pieces of information =
about each user:</div>
<div>&gt; 1. ID and credentials</div>
<div>&gt; 2. User profile to control the level of service provided to the u=
ser.</div>
</div>
<div><br>
</div>
</span><div>What is your app server using that ID and credentials for, if n=
ot to compare to something that is effectively an ACL - like, to prevent Al=
ice from getting Bob&#39;s voicemail? My point earlier was that if all you =
care about is showing that someone is Alice
 or Bob (or just that they&#39;re an authenticated user and it doesn&#39;t =
matter who), then a solution to the requirements could look very similar to=
 RFC4474 or STIR: you just need authentication, not authorization, and call=
ing it authorization is just confusing the
 issue.</div>
<div><br></div></div></blockquote><div><br></div><div>I care about both, au=
thentication and authorization, and my argument is that an ID Token could b=
e used to provide the needed user authentication, and the scope associated =
with the ID Token could be used to provide the authorization to control acc=
ess to services for that specific user.</div><div><br></div><div>=A0</div><=
blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-l=
eft-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;pa=
dding-left:1ex"><div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-si=
ze:14px;font-family:Calibri,sans-serif"><div>
</div>
<div>But, there&#39;s a different approach here, one where a service doesn&=
#39;t need to know IDs or credentials or anything about users. </div></div>=
</blockquote><div><br></div><div>In this specific case, the service obvious=
ly needs to know the user ID, but it does not have to know the user&#39;s c=
redentials to authorize access to the voicemail box.</div><div><br></div><d=
iv>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0=
.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-s=
tyle:solid;padding-left:1ex"><div style=3D"word-wrap:break-word;color:rgb(0=
,0,0);font-size:14px;font-family:Calibri,sans-serif"><div>This is an approa=
ch where I just get a token from an authorization service, and hand that to=
ken to the app service, and the app
 service uses that token *instead of* my ID or credentials or any personal =
profile about me. </div></div></blockquote><div><br></div><div>As I mention=
ed above, the ID Token proves your identity and has a scope that controls y=
our access to services.</div><div><br></div><div>=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;bo=
rder-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">=
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif"><div>I can imagine a very odd voicemail service tha=
t, rather than putting messages into buckets for Alice and Bob, put them al=
l into one generic bucket - and then an outside
 authorization service kept track of the fact that messages 2, 17, and 109 =
are for Alice. When Alice requests access to her voicemail from the authori=
zation service, it gives her a token that proves to the voicemail service t=
hat she is allowed to access only
 messages 2, 17, and 109, and she can only do it in the next five minutes, =
say, to prevent capture and replay. The voicemail service then has no idea =
who she is or who those messages are for, it just ponies them up. If *that*=
 is what you said above, at least
 I would understand why you want to use OAuth - it would be a terrible arch=
itecture (the authZ service keeps track of those messages how exactly?), bu=
t at least this is what OAuth is for. The architectures make more sense wit=
h third parties.</div>
<div><br></div></div></blockquote><div><br></div><div>That obviously does n=
ot make sense for a voicemail service,<br></div><div><br></div><div>But, le=
t&#39;s take a look at a different example; an adhoc conference server:</di=
v><div>I can imagine an adhoc conference server that accepts requests to al=
locate a focus to a user based on the ID Token provided in the incoming req=
uest. The ID Token proves the identity of the user and provides a scope to =
control the level of service provided to the user. The conference server wi=
ll be able to verify the validity of the ID Token because the token has the=
 signature of an Authorization Server that the conference server trusts.</d=
iv><div>In this specific case, do you see a need for the conference server =
to maintain a user directory?</div><div><br></div><div>In any case, I am no=
t trying to completely get rid of user directory from all services, but mai=
nly to get rid of the need to maintain separate user identities and credent=
ials per service.</div><div>Ideally, I would like the user to have one iden=
tity and be able to use that identity to authenticate and get access to all=
 services in the enterprise.</div><div><br></div><div><br></div><blockquote=
 class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:=
1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left=
:1ex"><div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;fo=
nt-family:Calibri,sans-serif"><div>
</div>
<div>A third approach would be problems we looked at as &quot;trait-based&q=
uot; authorization problems. The classic example is where for some service,=
 perhaps it matters whether Alice is in the Engineering department or the F=
inance department, rather than who Alice is
 exactly. Maybe those&#39;re the sort of elements you mean by &quot;user pr=
ofile.&quot; We did a whole requirements process years ago on &quot;trait-b=
ased&quot; authorization mechanisms intended to satisfy cases like this: se=
e RFC4484. You&#39;ll note it shows architectures where a SIP
 endpoints point by-reference to HTTP objects that contain authorization po=
licy elements like such &quot;traits.&quot; There are a bunch of solutions =
for that space, too, but if that&#39;s what you want, then my question for =
you would be, do you have any requirements beyond
 what&#39;s defined in RFC4484 Sec. 5?</div>
<div><br></div></div></blockquote><div><br></div><div>I have never read RFC=
4484 before; I will take a look.</div><div><br></div><div>Regards,</div><di=
v>=A0Rifaat</div><div><br></div><div><br></div><div><br></div><div>=A0</div=
><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border=
-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;=
padding-left:1ex"><div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-=
size:14px;font-family:Calibri,sans-serif"><div>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
<div><br>
</div>
<span>
<div style=3D"font-family:Calibri;font-size:11pt;text-align:left;color:blac=
k;border-width:1pt medium medium;border-style:solid none none;padding:3pt 0=
in 0in;border-top-color:rgb(181,196,223)">
<span style=3D"font-weight:bold">From: </span>Rifaat Shekh-Yusef &lt;<a hre=
f=3D"mailto:rifaat.ietf@gmail.com" target=3D"_blank">rifaat.ietf@gmail.com<=
/a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Wednesday, September 17, 2014=
 at 6:22 AM<div><div><br>
<span style=3D"font-weight:bold">To: </span>Jon Peterson &lt;<a href=3D"mai=
lto:jon.peterson@neustar.biz" target=3D"_blank">jon.peterson@neustar.biz</a=
>&gt;<br>
<span style=3D"font-weight:bold">Cc: </span>Paul Kyzivat &lt;<a href=3D"mai=
lto:pkyzivat@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit.edu</a>&gt;,=
 &quot;<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.o=
rg</a>&quot; &lt;<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipc=
ore@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [sipcore] A SIP OAuth =
use case<br>
</div></div></div><div><div>
<div><br>
</div>
<div>
<div>
<div dir=3D"ltr"><br>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Mon, Sep 15, 2014 at 3:29 PM, Peterson, Jon <=
span dir=3D"ltr">
&lt;<a href=3D"mailto:jon.peterson@neustar.biz" target=3D"_blank">jon.peter=
son@neustar.biz</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div><br>
</div>
<div>Format snags aside, </div>
</div>
</blockquote>
<div><br>
</div>
<div>I copied that text from a different editor.</div>
<div>The format seems ok here:</div>
<div><a href=3D"http://www.ietf.org/mail-archive/web/sipcore/current/msg062=
58.html" target=3D"_blank">http://www.ietf.org/mail-archive/web/sipcore/cur=
rent/msg06258.html</a><br>
</div>
<div><br>
</div>
<div>So, it might be something on your side?</div>
<div><br>
</div>
<div>=A0<br>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div>I have a bit of difficulty even differentiating the issues listed belo=
w: surely the reason why enterprises assign multiple identities per user is=
 to authorize access to different services separately.
</div>
</div>
</blockquote>
<div><br>
</div>
<div>Sorry, I do not get that.=A0</div>
<div>Why would the enterprise be required to create multiple identities to =
allow access to different services separately? in other words, why cannot t=
he enterprise create one identity and still be able to control the user&#39=
;s access to different services provided
 by different servers?<br>
</div>
<div><br>
</div>
<div>=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div>This is an issue that could be resolved with an access control list pe=
r service and a trusted identity broker in the enterprise vouching for user=
 identities to these services.
</div>
</div>
</blockquote>
<div><br>
</div>
<div>But that means that each server that provides a service must maintain =
a separate user directory, which is exactly what I am trying to avoid.</div=
>
<div><br>
</div>
<div>=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div>But to convince ourselves that one approach to this is better than ano=
ther, we need to look at a level of detail below these issue descriptions. =
Again, in order to access a voicemail service in an enterprise, say, what d=
oes the voicemail service need to
 know about the user?</div>
<div><br>
</div>
<div>Like, for regular access to the voicemail for &quot;jon@enterprise,&qu=
ot; surely all the voicemail service needs to know is that the entity autho=
ritative for the @enterprise issued me the name &quot;jon.&quot; Maybe the =
&quot;admin&quot; account gets to access all the voicemails, if
 need be. </div>
</div>
</blockquote>
<div><br>
</div>
<div>
<div>The application server needs the following pieces of information about=
 each user:</div>
<div>1. ID and credentials</div>
<div>2. User profile to control the level of service provided to the user.<=
/div>
</div>
<div><br>
</div>
<div>You cannot assume that all users will get the same level of service.</=
div>
<div><br>
</div>
<div>=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div>There are a variety of mechanisms that suffice to prove simple identit=
ies like that to a service.
</div>
</div>
</blockquote>
<div><br>
</div>
<div>Can you elaborate?</div>
<div>Are you assuming that the service still have a separate user directory=
?</div>
<div><br>
</div>
<div>=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div>If, however, I as &quot;jon&quot; needed for a third-party transcripti=
on service to access a subset of the messages in my voicemail box, and I wa=
nted the permissions for that access to expire as soon as the messages had =
been transcribed, then there is a much narrower
 set of mechanisms that could provide that capability. </div>
</div>
</blockquote>
<div><br>
</div>
<div>You keep going back to the OAuth 2.0 model and try to impose it on SIP=
, which would work in some use cases.</div>
<div><br>
</div>
<div>What I am proposing is something along the lines of OpenID, which exte=
nds the OAuth to authenticate the *user*.</div>
<div>In this case, the mapping between OAuth and SIP would be something lik=
e the following:</div>
<div><br>
</div>
<div>OAuth =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0SIP</div>
<div>----------------------------------------------------------------------=
-----</div>
<div>Resource Owner =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 End User</div>
<div>Resource Server =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 SIP Proxy or S=
IP Application Server</div>
<div>Client =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 =A0 SIP UA</div>
<div>Authorization Server =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Authorization Serv=
er</div>
<div><br>
</div>
<div>Do you see a problem with this model?</div>
<div><br>
</div>
<div>=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div>But it sounds to me, from this issues list, like your requirements are=
 more along the former lines than the latter.=A0</div>
<div><br>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>Actually, it is both.</div>
<div><br>
</div>
<div>Here is an example of a use case that fits with the OAuth 2.0 model:</=
div>
<div><br>
</div>
<div>An Application Server that provides Mobile Clients with access to SIP =
services.</div>
<div>The Mobile Client uses a proprietary protocol to communicate with the =
Application Server that registers and gets service from the SIP network on =
behalf of the user that is using the Mobile Client.</div>
<div><br>
</div>
<div>The communication channels between the players in this case are as fol=
lows (hopefully the format is ok this time):</div>
<div><br>
</div>
<div>[Mobile Client] =A0&lt;-----------&gt; =A0[Application Server] &lt;---=
--------&gt; [SIP Proxy]</div>
<div>=A0 =A0 =A0 =A0 =A0 /\ =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 /\</div>
<div>=A0 =A0 =A0 =A0 =A0 =A0| =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|</div>
<div>=A0 =A0 =A0 =A0 =A0 =A0| =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 \/<br>
</div>
<div>=A0 =A0 =A0 =A0 =A0 =A0------------------------&gt; [Authorization Ser=
ver]</div>
<div><br>
</div>
<div><br>
</div>
<div>In this case, the mapping between OAuth and SIP would be something lik=
e the following:<br>
</div>
<div>
<div><br>
</div>
<div>OAuth =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0SIP</div>
<div>----------------------------------------------------------------------=
-----</div>
<div>Resource Owner =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 End User</div>
<div>Resource Server =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 SIP Proxy</div=
>
<div>Client =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 =A0 Application Server (providing service to the mobile client)</div>
<div>Authorization Server =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Authorization Serv=
er</div>
</div>
<div><br>
</div>
<div><br>
</div>
<div>Regards,</div>
<div>=A0Rifaat</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div>=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div></div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
<div><br>
</div>
<span>
<div style=3D"font-family:Calibri;font-size:11pt;text-align:left;color:blac=
k;border-width:1pt medium medium;border-style:solid none none;padding:3pt 0=
in 0in;border-top-color:rgb(181,196,223)">
<span style=3D"font-weight:bold">From: </span>Rifaat Shekh-Yusef &lt;<a hre=
f=3D"mailto:rifaat.ietf@gmail.com" target=3D"_blank">rifaat.ietf@gmail.com<=
/a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Saturday, September 13, 2014 =
at 12:32 PM
<div>
<div><br>
<span style=3D"font-weight:bold">To: </span>Jon Peterson &lt;<a href=3D"mai=
lto:jon.peterson@neustar.biz" target=3D"_blank">jon.peterson@neustar.biz</a=
>&gt;<br>
<span style=3D"font-weight:bold">Cc: </span>Paul Kyzivat &lt;<a href=3D"mai=
lto:pkyzivat@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit.edu</a>&gt;,=
 &quot;<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.o=
rg</a>&quot; &lt;<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipc=
ore@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [sipcore] A SIP OAuth =
use case<br>
</div>
</div>
</div>
<div>
<div>
<div><br>
</div>
<div>
<div>
<div dir=3D"ltr"><br>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Thu, Sep 4, 2014 at 7:27 PM, Peterson, Jon <s=
pan dir=3D"ltr">
&lt;<a href=3D"mailto:jon.peterson@neustar.biz" target=3D"_blank">jon.peter=
son@neustar.biz</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div><br>
</div>
<div>I understand that OAuth can be altered from its basic flows to provide=
 tokens for OpenID, and IdP could be bound to OAuth, and while this is all =
very interesting it doesn&#39;t change much about the requirements discussi=
on I think we need to have: the one
 about what kinds of use cases we want to satisfy and who the relying parti=
es are in those cases. Once we understand that, we will understand what too=
ls are applicable to these use cases - preferably, not tools that need to b=
e twisted into a different shape
 to be applicable.</div>
<div><br>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>Well, OpenID extends the OAuth 2.0 Framework to authenticate the user;=
 why do you think it is &quot;twisting&quot; OAuth into a different shape?<=
/div>
<div><br>
</div>
<div><br>
</div>
<div>=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div></div>
<div>It would helpful, in the enterprise SSO case, for us to understand wha=
t kinds of application servers a user might need to access, and what exactl=
y those application servers need to know about the user in order to accompl=
ish their function.</div>
<div><br>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>Below is a description for some enterprise issues, that hopefully will=
 be addressed by a new authorization framework:</div>
<div><br>
</div>
<div><br>
</div>
<div><span style=3D"color:rgb(0,0,0);font-family:&#39;Courier New&#39;,Cour=
ier,monospace;font-size:14px;white-space:pre-wrap">* Multiple Identities On=
e of the major issues in the enterprise is the need for a user to obtain mu=
ltiple uncorrelated identities
 associated with different products to get access to the various services p=
rovided by the enterprise. For example, in the enterprise the user will be =
provided with different identities to: o Login his device (PC, laptop, mobi=
le, ...) to the network. o Register
 to the SIP network and get basic SIP services. o Access his Voice Mail. o =
Host a conference. o etc Typically, these identities belong to different ad=
ministrative domains and realms. Instead of these multiple identities, idea=
lly the enterprise would like to
 create and maintain only one identity for the user and then allow the user=
 access to all SIP and non-SIP services using that one identity. * Authoriz=
ation of Access to Services Another issue in the enterprise is the need to =
control and authorize a user access
 to services. Because application servers have their own user directory, th=
e control of access to services is spread over a multitude of servers in th=
e network. Ideally, the enterprise would like to manage the access to servi=
ces using one centralized entity
 and for the various entities that provide the service to become the enforc=
ement points. * Assignment of Services In the enterprise, different users m=
ight be assigned to different servers providing a specific service; e.g. Vo=
ice Mail Box for different users
 might be hosted by different servers, and the user need to know his assign=
ed server to be able to get access to this service. * Application Servers T=
o be able to provide service, the application server needs a list of users =
allowed to get services. The application
 server needs the following pieces of information about each user: 1. User =
ID and Credentials 2. Profile to control the level of service provided to t=
he user. If one application server provides different services using differ=
ent protocols, the application server
 might need to maintain different identities associated with these protocol=
s; e.g. an application server that provides presence service using SIP, and=
 IM service using XMPP. Ideally, an authorization framework would: 1. Off l=
oad the application server from
 the need to manage the user credentials and authenticating the user. 2. Of=
f load the application server from the need to maintain the users and their=
 profiles to control the level of service provided to the user. 3. Allow th=
e assignment of a user to a specific
 application server. * OAuth/OpenID Model With the OAuth/OpenID framework, =
the user ID and credentials will be handled by the authorization server, an=
d the application server assignment and the level of service associated wit=
h the user will be provided in the
 scope of the token associated with the user. When the application server r=
eceives a request from a user it will have all the details that would allow=
 the application server to validate the request to make sure it is coming f=
rom the right authorization server,
 and to act upon the request because it has the scope associated with the u=
ser, without the need to contact the authorization server. * Relying Party =
Regarding the Relying Party question: I think that when the user is being a=
uthenticated, that the Relying Party
 would be the SIP Client that would be given a user token (ID Token in Open=
ID lingo) to access services on behalf of the end user. So, the mapping wou=
ld be something like this: +------------------------+----------------------=
-------+ | OAuth | SIP | +------------------------+------------------------=
-----+
 | Resource Owner | End User | | Resource Server | SIP Proxy or SIP App Ser=
ver | | Client | SIP UA (Relying Party) | | Authorization Server | Authoriz=
ation Server | +------------------------+-----------------------------+</sp=
an><br>
</div>
<div><span style=3D"color:rgb(0,0,0);font-family:&#39;Courier New&#39;,Cour=
ier,monospace;font-size:14px;white-space:pre-wrap"><br>
</span></div>
<div><br>
</div>
<div>Regards,</div>
<div>=A0Rifaat</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div>=A0<br>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div></div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
<div><br>
</div>
<span>
<div style=3D"font-family:Calibri;font-size:11pt;text-align:left;color:blac=
k;border-width:1pt medium medium;border-style:solid none none;padding:3pt 0=
in 0in;border-top-color:rgb(181,196,223)">
<span style=3D"font-weight:bold">From: </span>Rifaat Shekh-Yusef &lt;<a hre=
f=3D"mailto:rifaat.ietf@gmail.com" target=3D"_blank">rifaat.ietf@gmail.com<=
/a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Thursday, September 4, 2014 a=
t 8:31 AM<br>
<span style=3D"font-weight:bold">To: </span>Jon Peterson &lt;<a href=3D"mai=
lto:jon.peterson@neustar.biz" target=3D"_blank">jon.peterson@neustar.biz</a=
>&gt;<br>
<span style=3D"font-weight:bold">Cc: </span>Paul Kyzivat &lt;<a href=3D"mai=
lto:pkyzivat@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit.edu</a>&gt;,=
 &quot;<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.o=
rg</a>&quot; &lt;<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipc=
ore@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [sipcore] A SIP OAuth =
use case<br>
</div>
<div>
<div>
<div><br>
</div>
<div>
<div>
<div dir=3D"ltr"><br>
<div class=3D"gmail_extra"><br>
<br>
<div class=3D"gmail_quote">On Wed, Sep 3, 2014 at 4:00 PM, Peterson, Jon <s=
pan dir=3D"ltr">
&lt;<a href=3D"mailto:jon.peterson@neustar.biz" target=3D"_blank">jon.peter=
son@neustar.biz</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
<br>
While I would echo Paul&#39;s thoughts below about the plausibility of serv=
ice<br>
providers using this model to authorize recording, I also agree that<br>
third-party recording is an excellent example of a service architecture<br>
that requires an authorization mechanism like OAuth. An end user needs to<b=
r>
authorize a third party to participate in the session under certain<br>
constraints, and coordinate the association between the third party and<br>
the VoIP service. This example also (rightly, in my opinion) conducts the<b=
r>
majority of the flow in HTTP where it belongs.<br>
<br>
</blockquote>
<div><br>
</div>
<div>I agree with that.</div>
<div><br>
</div>
<div>=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
Many of the use cases under discussion in this thread, though, are of the<b=
r>
form where there are really only two involved parties: an end user and a<br=
>
service provider. If you are my service provider, then in traditional SIP<b=
r>
I share a secret with you that I use to REGISTER my devices. I don&#39;t th=
ink<br>
you need OAuth for that. </blockquote>
<div><br>
</div>
<div>If you assume that there is one server that provides all the services,=
 then you are right.</div>
<div>But that is not the case all the time. I think that Dale articulated t=
hat clearly in the following response:</div>
<div><a href=3D"http://www.ietf.org/mail-archive/web/sipcore/current/msg062=
33.html" target=3D"_blank">http://www.ietf.org/mail-archive/web/sipcore/cur=
rent/msg06233.html</a><br>
</div>
<div><br>
</div>
<div>=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
I do understand that OAuth 2.0 is used in the<br>
wild for SSO, though I sympathize with Matt that the base flow shown in<br>
RFC6749 Sec 1.2 just doesn&#39;t map onto the requirements for SSO.<br>
<br>
</blockquote>
<div><br>
</div>
<div>I agree that the OAuth 2.0 &quot;as is&quot; does not address the SSO =
requirements.</div>
<div>The SIP OAuth proposal is not just using OAuth 2.0 &quot;as is&quot;, =
but it is extending the framework to authenticate and provide a *user* spec=
ific token.</div>
<div><br>
</div>
<div>Since Matt mentioned OpenID few times, I looked at the OpenID specific=
ation.</div>
<div>What they are doing there is that they defined a new token, called ID =
Token, that is associated with the *user*; which is exactly what the SIP OA=
uth proposal is doing.</div>
<div><a href=3D"http://openid.net/specs/openid-connect-core-1_0.html#IDToke=
n" target=3D"_blank">http://openid.net/specs/openid-connect-core-1_0.html#I=
DToken</a><br>
</div>
<div><br>
</div>
<div>My plan is to use the same terminology, ID Token, in the next version =
of the SIP OAuth document to clarify that, and to differentiate it from the=
 access token.</div>
<div><br>
</div>
<div>=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
I think, though, we might usefully break uses cases here down by who the<br=
>
intended relying party is:<br>
<br>
- The user&#39;s own SIP service provider (my enterprise, my carrier, my OT=
T<br>
provider)<br>
- A third party that a user wants to authorize for a service (recording<br>
service... But anything else?)<br>
<br>
I would also append some use cases where the relying party:<br>
<br>
- Someone with no association with the user (caller ID sorts of cases)<br>
<br>
This last is where I&#39;ve argued things like IdP might be relevant.<br>
<br>
</blockquote>
<div><br>
</div>
<div>EKR has a nice description for binding IdP to OAuth in his WebRTC Secu=
rity Architecture document here:</div>
<div><a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-security-arch=
-10#appendix-A.2" target=3D"_blank">https://tools.ietf.org/html/draft-ietf-=
rtcweb-security-arch-10#appendix-A.2</a><br>
</div>
<div><br>
</div>
<div>We could use the same idea for SIP.</div>
<div><br>
</div>
<div><br>
</div>
<div>Regards,</div>
<div>=A0Rifaat</div>
<div><br>
</div>
<div><br>
</div>
<div>=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
Are there any other high-level buckets of use cases here?<br>
<span><font color=3D"#888888"><br>
Jon Peterson<br>
Neustar, Inc.<br>
</font></span>
<div>
<div><br>
On 8/25/14, 11:44 AM, &quot;Paul Kyzivat&quot; &lt;<a href=3D"mailto:pkyziv=
at@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit.edu</a>&gt; wrote:<br>
<br>
&gt;On 8/25/14 11:50 AM, Rifaat Shekh-Yusef wrote:<br>
&gt;&gt; Hi Matt,<br>
&gt;&gt;<br>
&gt;&gt; The use case and the solution provided assumes that the VoIP Provi=
der<br>
&gt;&gt; makes the decision on when and what to record.<br>
&gt;&gt; While this is a valid use case, there are other use cases that all=
ow the<br>
&gt;&gt; user to decide on when and what to record; take a look at RFC6341 =
for<br>
&gt;&gt; more info.<br>
&gt;&gt; <a href=3D"http://datatracker.ietf.org/doc/rfc6341/" target=3D"_bl=
ank">http://datatracker.ietf.org/doc/rfc6341/</a><br>
&gt;<br>
&gt;SIPREC supports numerous topologies.<br>
&gt;It would support the one that Matt outlines as well as end-user<br>
&gt;controlled ones.<br>
&gt;<br>
&gt;I found Matt&#39;s use case interesting. I think it would be nice if su=
ch<br>
&gt;services were available in that form. But I have difficulty imagining<b=
r>
&gt;any of the VoIP providers I&#39;m aware of supporting this model. My<br=
>
&gt;impression is that this means that they are giving up a value added<br>
&gt;revenue market, that they are loath to do. The most likely justificatio=
n<br>
&gt;I can imagine that they don&#39;t want to legal consequences of doing<b=
r>
&gt;recording.<br>
&gt;<br>
&gt;=A0 =A0 =A0 =A0Thanks,<br>
&gt;=A0 =A0 =A0 =A0Paul<br>
&gt;<br>
&gt;&gt; For these use cases, where the user is in control, you need to awa=
y to<br>
&gt;&gt; authenticate and control which users are allowed to record and the=
 level<br>
&gt;&gt; of service provided for that specific user.<br>
&gt;&gt; This is where the solutions provided in section 3 or section 4 of =
the<br>
&gt;&gt; SIP OAuth proposal come into play.<br>
&gt;&gt; Obviously, in this case the authorization server is different from=
<br>
&gt;&gt; authorization server used to establish the trust relationship betw=
een<br>
&gt;&gt; the VoIP Provider and the CRS Provider.<br>
&gt;&gt; In this case, the authorization server must authenticate the user =
and<br>
&gt;&gt; provide his UA with an access token or a code that controls the us=
er&#39;s<br>
&gt;&gt; access to the service.<br>
&gt;&gt;<br>
&gt;&gt; Regards,<br>
&gt;&gt;=A0 =A0Rifaat<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; On Sat, Aug 23, 2014 at 9:05 AM, Matt Ryan &lt;<a href=3D"mailto:i=
etf@mvryan.org" target=3D"_blank">ietf@mvryan.org</a><br>
&gt;&gt; &lt;mailto:<a href=3D"mailto:ietf@mvryan.org" target=3D"_blank">ie=
tf@mvryan.org</a>&gt;&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A0Included is the use case I spoke of before.=A0 My apolog=
ies for the<br>
&gt;&gt;delay.<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A0This is written as though it could be included in<br>
&gt;&gt;=A0 =A0 =A0[draft-yusef-sipcore-sip-oauth] at section 6.<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A06. Examples<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A06.1. Example: Call Recording Service<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A0This example illustrates the use of SIP OAuth between th=
ree<br>
&gt;&gt;=A0 =A0 =A0parties:=A0 A hosted VoIP service provider, a Call Recor=
ding Service<br>
&gt;&gt;=A0 =A0 =A0provider, and a phone system end-user.=A0 In this exampl=
e:<br>
&gt;&gt;=A0 =A0 =A0- The phone system end-user is a customer of both the ho=
sted VoIP<br>
&gt;&gt;=A0 =A0 =A0provider and the Call Recording Service provider<br>
&gt;&gt;=A0 =A0 =A0- The hosted VoIP service provider is a standard provide=
r of hosted<br>
&gt;&gt;=A0 =A0 =A0business-class VoIP telephone service using SIP<br>
&gt;&gt;=A0 =A0 =A0- The Call Recording Service provider is a distinct enti=
ty from the<br>
&gt;&gt;=A0 =A0 =A0hosted VoIP provider<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A0The Call Recording Service provides to customers both th=
e call<br>
&gt;&gt;=A0 =A0 =A0recording abilities and management of call recordings<br=
>
&gt;&gt;=A0 =A0 =A0(configuration, sharing and distribution, retention, etc=
.).=A0 This<br>
&gt;&gt;=A0 =A0 =A0service can accept and record RTP traffic, associate all=
 RTP streams<br>
&gt;&gt;=A0 =A0 =A0with a single call together, and store and catalog the r=
ecorded data<br>
&gt;&gt;=A0 =A0 =A0for future searching and retrieval.=A0 The service also =
offers a web<br>
&gt;&gt;=A0 =A0 =A0interface where customers may view and retrieve stored c=
alls.<br>
&gt;&gt;=A0 =A0 =A0Stored calls may range from simple person-to-person voic=
e calls to<br>
&gt;&gt;=A0 =A0 =A0multi-party conferences with a multitude of audio and vi=
deo<br>
&gt;&gt;=A0 =A0 =A0streams.=A0 In this example, customers are billed based =
on the amount<br>
&gt;&gt;=A0 =A0 =A0of data they store, although other models are certainly =
possible.<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A06.1.1. OAuth 2.0 Roles<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A0In this example, each party assumes the following OAuth =
2.0 roles as<br>
&gt;&gt;=A0 =A0 =A0defined in [RFC6749] Section 1.1:<br>
&gt;&gt;=A0 =A0 =A0- The end-user assumes the role of resource owner<br>
&gt;&gt;=A0 =A0 =A0- The hosted VoIP service provider assumes the role of c=
lient<br>
&gt;&gt;=A0 =A0 =A0- The Call Recording Service provider assumes the role o=
f resource<br>
&gt;&gt;=A0 =A0 =A0server as well as the role of authorization server<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A06.1.2. Description<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A0There are two parts to the example:=A0 Initial configura=
tion and<br>
&gt;&gt;=A0 =A0 =A0real-time recording authorization.<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A0Each section includes a simplified flow diagram for the =
purpose of<br>
&gt;&gt;=A0 =A0 =A0describing the corresponding portion of the example.=A0 =
For the sake<br>
&gt;&gt;=A0 =A0 =A0of brevity, certain parts of the OAuth dialog have been =
eliminated.<br>
&gt;&gt;=A0 =A0 =A0Implements should refer to [RFC6749] for details on OAut=
h.<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A06.1.2.1 Initial Configuration<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A0 =A0 +-------------+=A0 =A0 =A0+---------------+=A0 =A0 =
=A0+--------------+<br>
&gt;&gt;=A0 =A0 =A0 =A0 |=A0 End-User=A0 =A0|=A0 =A0 =A0| VoIP Provider |=
=A0 =A0 =A0| CRS Provider |<br>
&gt;&gt;=A0 =A0 =A0 =A0 | Web Browser |=A0 =A0 =A0|=A0 =A0 Website=A0 =A0 |=
=A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 |<br>
&gt;&gt;=A0 =A0 =A0 =A0 +-------------+=A0 =A0 =A0+---------------+=A0 =A0 =
=A0+--------------+<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 |=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| HTTP 1=A0 =A0 =A0 =A0 =A0 =A0 =A0|=
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| (Configure CRS)=A0 =A0 |=A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|-------------------&gt;|=A0 =A0 =A0=
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 |=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| HTTP 2=A0 =A0 =A0 =A0 =A0 =A0 =A0|=
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| (OAuth Redirect=A0 =A0 |=A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 to CRS Website)=A0 =A0|=A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|&lt;-------------------|=A0 =A0 =A0=
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 |=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| HTTP 3=A0 =A0 =A0 =A0 =A0 =A0 =A0|=
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| (Authorize SIP from VoIP provider=
=A0 =A0 =A0 =A0 |<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|-----------------------------------=
------&gt;|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 |=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| HTTP 4=A0 =A0 =A0 =A0 =A0 =A0 =A0|=
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| (OAuth Redirect back to VoIP porta=
l)=A0 =A0 =A0|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|&lt;-------------------------------=
----------|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 |=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| HTTP 5=A0 =A0 =A0 =A0 =A0 =A0 =A0|=
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|-------------------&gt;|=A0 =A0 =A0=
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 | HTTP 6=A0 =A0 =A0 =A0 =A0 =A0 =A0 |<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 | (Request Access and |<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 |=A0 refresh tokens)=A0 =A0 |<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 |--------------------&gt;|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 |=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A0Some time after having signed up for both services, but =
prior to<br>
&gt;&gt;=A0 =A0 =A0attempting to record any calls, the end-user logs into t=
he web<br>
&gt;&gt;=A0 =A0 =A0portal of the hosted VoIP provider with a web browser an=
d configures<br>
&gt;&gt;=A0 =A0 =A0their call recording service (HTTP 1).=A0 This configura=
tion includes<br>
&gt;&gt;=A0 =A0 =A0providing a URI where the call recording service may be =
reached,<br>
&gt;&gt;=A0 =A0 =A0along with a set of API credentials, provided by the cal=
l recording<br>
&gt;&gt;=A0 =A0 =A0service, which will allow the client to initiate an OAut=
h exchange.<br>
&gt;&gt;=A0 =A0 =A0The end-user also configures the conditions under which =
call<br>
&gt;&gt;=A0 =A0 =A0recordings should take place.=A0 For example, the end-us=
er may request<br>
&gt;&gt;=A0 =A0 =A0to record every multi-party (conference) call, or every =
call<br>
&gt;&gt;=A0 =A0 =A0involving a specific set of users.=A0 The end-user may a=
lso specify<br>
&gt;&gt;=A0 =A0 =A0other restrictions, such as restricting codec negotiatio=
n to G.711u<br>
&gt;&gt;=A0 =A0 =A0for audio and H.264 for video in order to record the RTP=
 streams.<br>
&gt;&gt;=A0 =A0 =A0Once the end-user saves the configuration, the hosted Vo=
IP provider<br>
&gt;&gt;=A0 =A0 =A0web portal redirects the end-user&#39;s web browser to t=
he call<br>
&gt;&gt;=A0 =A0 =A0recording service website and a standard HTTP OAuth dial=
og begins<br>
&gt;&gt;=A0 =A0 =A0(HTTP 2).=A0 The end-user authorizes the hosted VoIP pro=
vider to<br>
&gt;&gt;=A0 =A0 =A0access (i.e. send SIP and RTP traffic to) the call recor=
ding<br>
&gt;&gt;=A0 =A0 =A0service, for the purpose of recording calls, so long as =
the<br>
&gt;&gt;=A0 =A0 =A0configured conditions are met (HTTP 3).=A0 The result of=
 this<br>
&gt;&gt;=A0 =A0 =A0interaction is that the hosted VoIP provider ends up rec=
eiving an<br>
&gt;&gt;=A0 =A0 =A0OAuth access token and refresh token from the call recor=
ding service<br>
&gt;&gt;=A0 =A0 =A0(HTTP 4, HTTP 5, HTTP 6).=A0 These tokens are saved in t=
he end-user&#39;s<br>
&gt;&gt;=A0 =A0 =A0hosted VoIP account database.<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A06.1.2.2 Real-time Recording Authorization<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A0 =A0 +-------------+=A0 =A0 =A0+---------------+=A0 =A0 =
=A0 +--------------+<br>
&gt;&gt;=A0 =A0 =A0 =A0 |=A0 End-User=A0 =A0|=A0 =A0 =A0| VoIP Provider |=
=A0 =A0 =A0 | CRS Provider |<br>
&gt;&gt;=A0 =A0 =A0 =A0 |=A0 SIP Phone=A0 |=A0 =A0 =A0|=A0 =A0 Website=A0 =
=A0 |=A0 =A0 =A0 |=A0 =A0 =A0 =A0 =A0 =A0 =A0 |<br>
&gt;&gt;=A0 =A0 =A0 =A0 +-------------+=A0 =A0 =A0+---------------+=A0 =A0 =
=A0 +--------------+<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 |=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| SIP INVITE 1=A0 =A0 =A0 =A0|=A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|-------------------&gt;|=A0 =A0 =A0=
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 | SIP INVITE 2=A0 =A0 =A0 =A0 =A0|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 | (with access token)=A0 |<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 |---------------------&gt;|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 |=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 | SIP 401 Unauthorized |<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 |&lt;---------------------|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 |=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 | SIP INVITE 3=A0 =A0 =A0 =A0 =A0|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 | (with refresh token) |<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 |---------------------&gt;|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 |=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 | SIP 200 1=A0 =A0 =A0 =A0 =A0 =A0 |<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 | (new access token)=A0 =A0|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 |&lt;---------------------|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 |=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 | SIP INVITE 4=A0 =A0 =A0 =A0 =A0|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 | (with access token)=A0 |<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 |---------------------&gt;|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 |=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 | SIP 200 2=A0 =A0 =A0 =A0 =A0 =A0 |<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 |&lt;---------------------|<br>
&gt;&gt;=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 |=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A0Independently of and after the initial configuration pha=
se, the<br>
&gt;&gt;=A0 =A0 =A0end-user makes a telephone call using the hosted VoIP pr=
ovider (SIP<br>
&gt;&gt;=A0 =A0 =A0INVITE 1).=A0 When this call takes place, the hosted VoI=
P provider<br>
&gt;&gt;=A0 =A0 =A0looks to see whether it believes this call should be rec=
orded.=A0 If<br>
&gt;&gt;=A0 =A0 =A0so, at this point it would branch the call session to th=
e call<br>
&gt;&gt;=A0 =A0 =A0recording service, using SIP OAuth to pass the previousl=
y provided<br>
&gt;&gt;=A0 =A0 =A0access token for authorization (SIP INVITE 2).=A0 Once t=
he access<br>
&gt;&gt;=A0 =A0 =A0token is validated by the call recording service (perhap=
s after<br>
&gt;&gt;=A0 =A0 =A0first providing a new access token based on receipt of a=
 valid<br>
&gt;&gt;=A0 =A0 =A0refresh token), the call recording service will check th=
e rights<br>
&gt;&gt;=A0 =A0 =A0that were previously authorized and examine the SIP to d=
etermine<br>
&gt;&gt;=A0 =A0 =A0whether it can accept the call.=A0 If so, it completes t=
he<br>
&gt;&gt;=A0 =A0 =A0establishment of the session and begins receiving and re=
cording the<br>
&gt;&gt;=A0 =A0 =A0RTP stream(s) (SIP 200 OK).=A0 The call recording servic=
e provider<br>
&gt;&gt;=A0 =A0 =A0could also deny the request, either because the tokens a=
re invalid<br>
&gt;&gt;=A0 =A0 =A0or because the content of<br>
&gt;&gt;=A0 =A0 =A0 =A0the SIP invite does not match the previously authori=
zed<br>
&gt;&gt;=A0 =A0 =A0conditions, in which case a SIP 403 would be returned by=
 the call<br>
&gt;&gt;=A0 =A0 =A0recording service provider and the call would not be rec=
orded -<br>
&gt;&gt;=A0 =A0 =A0however, the initial call branch would continue without<=
br>
&gt;&gt;interruption.<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A0Note:<br>
&gt;&gt;=A0 =A0 =A0[RFC6749] specifies that when a client uses a refresh to=
ken to<br>
&gt;&gt;=A0 =A0 =A0request a new access token, the response is HTTP 200 wit=
h the new<br>
&gt;&gt;=A0 =A0 =A0access token and optionally a new refresh token included=
 in the<br>
&gt;&gt;=A0 =A0 =A0payload.=A0 In this example, a SIP 200 response (SIP 200=
 1) is sent<br>
&gt;&gt;=A0 =A0 =A0which contains the new token(s).=A0 However, this could =
be confusing<br>
&gt;&gt;=A0 =A0 =A0as SIP 200 is generally viewed as the acceptance of the =
INVITE,<br>
&gt;&gt;=A0 =A0 =A0which is not what is happening in this case.=A0 Should S=
IP 200 be used<br>
&gt;&gt;=A0 =A0 =A0for consistency, or should another mechanism be used?<br=
>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A06.1.3. Justification<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A06.1.3.1. Hosted VoIP Service Provider<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A0In this example, the hosted VoIP service provider can al=
low their<br>
&gt;&gt;=A0 =A0 =A0customers to enable call recording in their VoIP service=
 by<br>
&gt;&gt;=A0 =A0 =A0selecting from any of a number of supported call recordi=
ng service<br>
&gt;&gt;=A0 =A0 =A0providers.=A0 This allows the hosted VoIP service provid=
er to offer<br>
&gt;&gt;=A0 =A0 =A0this feature to customers without requiring that the hos=
ted VoIP<br>
&gt;&gt;=A0 =A0 =A0service provider implement and support this feature them=
selves.<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A06.1.3.2. Call Recording Service Provider<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A0In this example, the Call Recording Service provider can=
 focus on<br>
&gt;&gt;=A0 =A0 =A0value and innovation in the area of recording calls and =
managing<br>
&gt;&gt;=A0 =A0 =A0recorded calls without having to build a full-featured t=
elephony<br>
&gt;&gt;=A0 =A0 =A0offering.<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A06.1.3.3. Customer<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A0In this example, the customer has more flexibility in ch=
oosing a<br>
&gt;&gt;=A0 =A0 =A0complete solution that meets all of the customer needs.=
=A0 The<br>
&gt;&gt;=A0 =A0 =A0customer does not have to settle for a substandard call =
recording<br>
&gt;&gt;=A0 =A0 =A0service offering in order to obtain other features they =
seek in a<br>
&gt;&gt;=A0 =A0 =A0hosted VoIP provider, and vice versa.<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A06.1.4. Variants<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A0A simple variant of this example is one wherein one of t=
he services<br>
&gt;&gt;=A0 =A0 =A0(either the VoIP service or the call recording service) =
is managed<br>
&gt;&gt;=A0 =A0 =A0directly by the end-user, but the other is not.=A0 For e=
xample, the<br>
&gt;&gt;=A0 =A0 =A0end-user may wish to make use of an external hosted VoIP=
 service<br>
&gt;&gt;=A0 =A0 =A0provider, but may have business or legal reasons that fo=
rbid the<br>
&gt;&gt;=A0 =A0 =A0end-user from allowing a third party to store and manage=
 recorded<br>
&gt;&gt;=A0 =A0 =A0calls.=A0 The end-user could choose to set up their own =
call recording<br>
&gt;&gt;=A0 =A0 =A0service as described in this example, and use SIP OAuth =
to<br>
&gt;&gt;=A0 =A0 =A0facilitate interaction between the hosted VoIP service a=
nd the<br>
&gt;&gt;=A0 =A0 =A0end-user&#39;s own call recording service.<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A06.2. Other SIP OAuth examples<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A0Many other SIP-based services could leverage SIP OAuth t=
o provide<br>
&gt;&gt;=A0 =A0 =A0value-added service to an end-user of a hosted VoIP serv=
ice<br>
&gt;&gt;=A0 =A0 =A0provider.=A0 Some examples of these types of services ar=
e listed<br>
&gt;&gt;below.<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A0Voicemail service provider:=A0 The end-user configures t=
heir hosted<br>
&gt;&gt;=A0 =A0 =A0VoIP service provider to manage voicemail through a sepa=
rate<br>
&gt;&gt;=A0 =A0 =A0voicemail service provider, which chooses whether to sto=
re voicemail<br>
&gt;&gt;=A0 =A0 =A0based on existing quotas, Caller ID, etc.<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A0Conferencing service provider:=A0 A conferencing service=
 may wish to<br>
&gt;&gt;=A0 =A0 =A0bill customers in a more granular fashion, for example, =
based on the<br>
&gt;&gt;=A0 =A0 =A0number of participants and attendees in a conference, th=
e duration<br>
&gt;&gt;=A0 =A0 =A0of conference, whether video was also included, which co=
decs were<br>
&gt;&gt;=A0 =A0 =A0being used, etc. instead of a flat monthly rate.=A0 SIP =
OAuth would<br>
&gt;&gt;=A0 =A0 =A0facilitate metered authorization for a client&#39;s use =
of the service,<br>
&gt;&gt;=A0 =A0 =A0instead of all-or-nothing access.<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A0Call center service provider:=A0 A third party may provi=
de ring groups<br>
&gt;&gt;=A0 =A0 =A0or call queues for a hosted VoIP provider along with det=
ailed<br>
&gt;&gt;=A0 =A0 =A0reports and dashboards.=A0 The end-user uses OAuth over =
SIP to control<br>
&gt;&gt;=A0 =A0 =A0who can log into which queues or ring groups, etc.<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A0--<br>
&gt;&gt;=A0 =A0 =A0Matt Ryan<br>
&gt;&gt;=A0 =A0 =A0code slinger | <a href=3D"http://zoomulus.com" target=3D=
"_blank">zoomulus.com</a> &lt;<a href=3D"http://zoomulus.com" target=3D"_bl=
ank">http://zoomulus.com</a>&gt;<br>
&gt;&gt;=A0 =A0 =A0ietf at mvryan dot org<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;=A0 =A0 =A0_______________________________________________<br>
&gt;&gt;=A0 =A0 =A0sipcore mailing list<br>
&gt;&gt;=A0 =A0 =A0<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">si=
pcore@ietf.org</a> &lt;mailto:<a href=3D"mailto:sipcore@ietf.org" target=3D=
"_blank">sipcore@ietf.org</a>&gt;<br>
&gt;&gt;=A0 =A0 =A0<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore=
" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sipcore</a><br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; _______________________________________________<br>
&gt;&gt; sipcore mailing list<br>
&gt;&gt; <a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf=
.org</a><br>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=
=3D"_blank">https://www.ietf.org/mailman/listinfo/sipcore</a><br>
&gt;&gt;<br>
&gt;<br>
&gt;_______________________________________________<br>
&gt;sipcore mailing list<br>
&gt;<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org<=
/a><br>
&gt;<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/sipcore</a><br>
<br>
_______________________________________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a><=
br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
>https://www.ietf.org/mailman/listinfo/sipcore</a><br>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</div>
</div>
</div>
</span></div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</div>
</div>
</div>
</span></div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</div>
</div></div></span>
</div>

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

--001a11c36748b9b4b00504ef2d25--


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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Thread-Topic: [sipcore] A SIP OAuth use case
Thread-Index: AQHPvtLmZYbTlD+FuE+zMNKWASQKqZvhvfeAgAAwjgCADcUOgIABvEuAgAAP2QCADliiAIACroIAgAMzZQCAH5scAIAB3UgA//+52YA=
Date: Wed, 8 Oct 2014 23:18:51 +0000
Message-ID: <D05B134B.134812%jon.peterson@neustar.biz>
References: <10D4F263-37CF-4A74-8F23-EBA9E5010E02@mvryan.org> <CAGL6ep+sc7DKcSF5gXDcGtX5oVMt1-=1w9JMsa1OQJb1AtCrkw@mail.gmail.com> <53FB83FA.8010101@alum.mit.edu> <D020D401.12E169%jon.peterson@neustar.biz> <CAGL6epKSipp6PiHnsC7i9iQKdkss3v0y2CZH4T+zt0Bmh2WRxQ@mail.gmail.com> <D02E41DF.131993%jon.peterson@neustar.biz> <CAGL6epJrS7H+t-CmoW4--MZcsPQJ4em2g2ZiTradLLJEcJEs-A@mail.gmail.com> <D03C75E8.133DCF%jon.peterson@neustar.biz> <CAGL6epJV=00VSba5mUZ2Oi_VpQKQgMA1ddqeRv5aq3wT9fOQbg@mail.gmail.com> <D059A86A.134723%jon.peterson@neustar.biz> <CAGL6epJi=ES6v5d7+4Uf43J3ugndcohNGOCbcoS441=aqdizjw@mail.gmail.com>
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Subject: Re: [sipcore] A SIP OAuth use case
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--_000_D05B134B134812jonpetersonneustarbiz_
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> The authentication could be done by an Authorization Server and as a resu=
lt the SIP endpoint would be given an ID Token, as per
> OpenID, which proves the identity of the end user. The SIP endpoint will =
be able to use the ID Token to get access to the user's
> box. The voicemail server will be able to identify the user and verify th=
e validity of the ID Token because the token has the
> signature of an Authorization Server that the voicemail server trusts.

Right, this is the authentication property that mechanisms like RFC4474, or=
 STIR, and to some extend RTCWeb IdP provide. A trusted service gives a chu=
nk of signed material to an endpoint, which the endpoint embeds in SIP requ=
ests, and that application services would be able to identify the user and =
verify that the identity has been checked by the trusted service.

> The ID Token could also have a scope that controls the services and level=
 of service provided to the user.

This "scope" is the part that I need to understand better - whether that re=
duces to something like the RFC4484 requirements, or if you have something =
different in mind, per your statement below as well. Without that, I don't =
get what you need beyond authentication.

> I care about both, authentication and authorization, and my argument is t=
hat an ID Token could be used to provide the needed
> user authentication, and the scope associated with the ID Token could be =
used to provide the authorization to control access to
> services for that specific user.

Jon Peterson
Neustar, Inc.

--_000_D05B134B134812jonpetersonneustarbiz_
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<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
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e-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif;">
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div>
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>&gt; The authentication could be done by an Authorization Server and a=
s a result the SIP endpoint would be given an ID Token, as per</div>
</div>
</div>
</div>
</div>
</div>
</span>
<div>&gt; OpenID, which proves the identity of the end user. The SIP endpoi=
nt will be able to use the ID Token to get access to the user's&nbsp;</div>
<div>&gt; box. The voicemail server will be able to identify the user and v=
erify the validity of the ID Token because the token has the&nbsp;</div>
<div>&gt; signature of an Authorization Server that the voicemail server tr=
usts.</div>
<div><br>
</div>
<div>Right, this is the authentication property that mechanisms like RFC447=
4, or STIR, and to some extend RTCWeb IdP provide. A trusted service gives =
a chunk of signed material to an endpoint, which the endpoint embeds in SIP=
 requests, and that application
 services would be able to identify the user and verify that the identity h=
as been checked by the trusted service.</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<div>&gt; The ID Token could also have a scope that controls the services a=
nd level of service provided to the user.</div>
</div>
</div>
</div>
</span>
<div><br>
</div>
<div>This &quot;scope&quot; is the part that I need to understand better - =
whether that reduces to something like the RFC4484 requirements, or if you =
have something different in mind, per your statement below as well. Without=
 that, I don't get what you need beyond authentication.</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<div>&gt; I care about both, authentication and authorization, and my argum=
ent is that an ID Token could be used to provide the needed</div>
</div>
</div>
</div>
</span>
<div>&gt; user authentication, and the scope associated with the ID Token c=
ould be used to provide the authorization to control access to</div>
<div>&gt; services for that specific user.</div>
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
</body>
</html>

--_000_D05B134B134812jonpetersonneustarbiz_--


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References: <10D4F263-37CF-4A74-8F23-EBA9E5010E02@mvryan.org> <CAGL6ep+sc7DKcSF5gXDcGtX5oVMt1-=1w9JMsa1OQJb1AtCrkw@mail.gmail.com> <53FB83FA.8010101@alum.mit.edu> <D020D401.12E169%jon.peterson@neustar.biz> <CAGL6epKSipp6PiHnsC7i9iQKdkss3v0y2CZH4T+zt0Bmh2WRxQ@mail.gmail.com> <D02E41DF.131993%jon.peterson@neustar.biz> <CAGL6epJrS7H+t-CmoW4--MZcsPQJ4em2g2ZiTradLLJEcJEs-A@mail.gmail.com> <D03C75E8.133DCF%jon.peterson@neustar.biz> <CAGL6epJV=00VSba5mUZ2Oi_VpQKQgMA1ddqeRv5aq3wT9fOQbg@mail.gmail.com> <D059A86A.134723%jon.peterson@neustar.biz> <CAGL6epJi=ES6v5d7+4Uf43J3ugndcohNGOCbcoS441=aqdizjw@mail.gmail.com> <D05B134B.134812%jon.peterson@neustar.biz>
Date: Thu, 9 Oct 2014 15:28:03 -0400
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From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
To: "Peterson, Jon" <jon.peterson@neustar.biz>
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Subject: Re: [sipcore] A SIP OAuth use case
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--089e0117716f5fd7650505026e47
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On Wed, Oct 8, 2014 at 7:18 PM, Peterson, Jon <jon.peterson@neustar.biz>
wrote:

>
>     > The authentication could be done by an Authorization Server and as
> a result the SIP endpoint would be given an ID Token, as per
>    > OpenID, which proves the identity of the end user. The SIP endpoint
> will be able to use the ID Token to get access to the user's
> > box. The voicemail server will be able to identify the user and verify
> the validity of the ID Token because the token has the
> > signature of an Authorization Server that the voicemail server trusts.
>
>  Right, this is the authentication property that mechanisms like RFC4474,
> or STIR, and to some extend RTCWeb IdP provide.
>

I need to educate myself more about RFC4474/STIR and RTCWeb IdP.



>
A trusted service gives a chunk of signed material to an endpoint, which
> the endpoint embeds in SIP requests, and that application services would be
> able to identify the user and verify that the identity has been checked by
> the trusted service.
>

Exactly.
OpenID adds Authentication to the OAuth 2.0 Authorization Framework by
introducing the ID Token structure to authenticate the end user.



>
>  > The ID Token could also have a scope that controls the services and
> level of service provided to the user.
>
>  This "scope" is the part that I need to understand better - whether that
> reduces to something like the RFC4484 requirements, or if you have
> something different in mind, per your statement below as well. Without
> that, I don't get what you need beyond authentication.
>

I briefly looked at RFC4484, section 5 (requirements), and it seems like a
reasonable approach.
I think that the framework should be flexible enough to allow different
scopes and claims to be associated with the user, but leave the content as
implementation detail.

Now if I to use RFC4484 terminology, and if I understood RFC4484 correctly,
then the combination of an Assertion that proves the identity of the user,
and the Traits that are associated with the Assertion, should allow the
system to provide the user with only one identity and allow that user
access to a variety of SIP *and* non-SIP services in the network. Correct?

Regards,
 Rifaat




>
>  > I care about both, authentication and authorization, and my argument
> is that an ID Token could be used to provide the needed
>   > user authentication, and the scope associated with the ID Token could
> be used to provide the authorization to control access to
> > services for that specific user.
>
>  Jon Peterson
> Neustar, Inc.
>

--089e0117716f5fd7650505026e47
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Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Wed, Oct 8, 2014 at 7:18 PM, Peterson, Jon <span dir=3D"ltr">&lt;<a =
href=3D"mailto:jon.peterson@neustar.biz" target=3D"_blank">jon.peterson@neu=
star.biz</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(20=
4,204,204);border-left-style:solid;padding-left:1ex">



<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif"><span>
<div><br>
</div>
<span>
<div>
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>&gt; The authentication could be done by an Authorization Server and a=
s a result the SIP endpoint would be given an ID Token, as per</div>
</div>
</div>
</div>
</div>
</div>
</span>
<div>&gt; OpenID, which proves the identity of the end user. The SIP endpoi=
nt will be able to use the ID Token to get access to the user&#39;s=A0</div=
>
<div>&gt; box. The voicemail server will be able to identify the user and v=
erify the validity of the ID Token because the token has the=A0</div>
<div>&gt; signature of an Authorization Server that the voicemail server tr=
usts.</div>
<div><br>
</div>
</span><div>Right, this is the authentication property that mechanisms like=
 RFC4474, or STIR, and to some extend RTCWeb IdP provide.</div></div></bloc=
kquote><div><br></div><div>I need to educate myself more about RFC4474/STIR=
 and RTCWeb IdP.</div><div>=A0</div><div><br></div><blockquote class=3D"gma=
il_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-le=
ft-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div st=
yle=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-family:Cal=
ibri,sans-serif"><div><span style=3D"font-family:arial;font-size:small;colo=
r:rgb(34,34,34)">=A0</span></div></div></blockquote><blockquote class=3D"gm=
ail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-l=
eft-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div s=
tyle=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-family:Ca=
libri,sans-serif"><div>A trusted service gives a chunk of signed material t=
o an endpoint, which the endpoint embeds in SIP requests, and that applicat=
ion
 services would be able to identify the user and verify that the identity h=
as been checked by the trusted service.</div></div></blockquote><div><br></=
div><div>Exactly.</div><div>OpenID adds Authentication to the OAuth 2.0 Aut=
horization Framework by introducing the ID Token structure to authenticate =
the end user.</div><div><br></div><div>=A0<br></div><blockquote class=3D"gm=
ail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-l=
eft-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div s=
tyle=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-family:Ca=
libri,sans-serif"><span>
<span>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<div>&gt; The ID Token could also have a scope that controls the services a=
nd level of service provided to the user.</div>
</div>
</div>
</div>
</span>
<div><br>
</div>
</span><div>This &quot;scope&quot; is the part that I need to understand be=
tter - whether that reduces to something like the RFC4484 requirements, or =
if you have something different in mind, per your statement below as well. =
Without that, I don&#39;t get what you need beyond authentication.</div></d=
iv></blockquote><div><br></div><div>I briefly looked at RFC4484, section 5 =
(requirements), and it seems like a reasonable approach.<br></div><div>I th=
ink that the framework should be flexible enough to allow different scopes =
and claims to be associated with the user, but leave the content as impleme=
ntation detail.</div><div><br></div><div>Now if I to use RFC4484 terminolog=
y, and if I understood RFC4484 correctly, then the combination of an Assert=
ion that proves the identity of the user, and the Traits that are associate=
d with the=A0Assertion, should allow the system to provide the user with=A0=
only one identity and allow that user access to a variety of SIP <b>and</b>=
 non-SIP services in the network. Correct?</div><div><br></div><div>Regards=
,</div><div>=A0Rifaat</div><div><br></div><div><br></div><div>=A0<br></div>=
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-s=
ize:14px;font-family:Calibri,sans-serif"><span>
<span>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<div>&gt; I care about both, authentication and authorization, and my argum=
ent is that an ID Token could be used to provide the needed</div>
</div>
</div>
</div>
</span>
<div>&gt; user authentication, and the scope associated with the ID Token c=
ould be used to provide the authorization to control access to</div>
<div>&gt; services for that specific user.</div>
<div><br>
</div>
</span><span><font color=3D"#888888"><div>Jon Peterson</div>
<div>Neustar, Inc.</div>
</font></span></div>

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

--089e0117716f5fd7650505026e47--


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To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Thread-Topic: [sipcore] A SIP OAuth use case
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References: <10D4F263-37CF-4A74-8F23-EBA9E5010E02@mvryan.org> <CAGL6ep+sc7DKcSF5gXDcGtX5oVMt1-=1w9JMsa1OQJb1AtCrkw@mail.gmail.com> <53FB83FA.8010101@alum.mit.edu>	<D020D401.12E169%jon.peterson@neustar.biz> <CAGL6epKSipp6PiHnsC7i9iQKdkss3v0y2CZH4T+zt0Bmh2WRxQ@mail.gmail.com> <D02E41DF.131993%jon.peterson@neustar.biz> <CAGL6epJrS7H+t-CmoW4--MZcsPQJ4em2g2ZiTradLLJEcJEs-A@mail.gmail.com> <D03C75E8.133DCF%jon.peterson@neustar.biz> <CAGL6epJV=00VSba5mUZ2Oi_VpQKQgMA1ddqeRv5aq3wT9fOQbg@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D45BB30@ESESSMB209.ericsson.se> <CAGL6epKjVCEzBFGYMYGv42V7_+Ey5kY+wZhCdqLgqkkMUXmZBg@mail.gmail.com>
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Subject: Re: [sipcore] A SIP OAuth use case
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Hi,

>I am not clear on the flow of messages in your case.
>Can you please provide a high level diagram that describes that flow?

The user contacts the WWSF, and uses normal web credentials (e.g. username =
+ password) to request an access token from the WWSF/WAF, using HTTP. The u=
ser then sends then includes the access token in the SIP REGISTER request.

So, as far as the flow is concerned, I guess the main difference from your =
draft is that the user does not retrieve the access token using SIP, but us=
ing HTTP.

>Also, the token that is returned by the WAF to the SIP endpoint; is it >an=
 access token, as per OAuth 2.0, or a user token (ID Token), as per >OpenID=
?

An access token.

Regards,

Christer

Hi,

>What I am proposing is something along the lines of OpenID, which extends =
the OAuth to >authenticate the *user*.
>In this case, the mapping between OAuth and SIP would be something like th=
e following:
>
>OAuth                                    SIP
>--------------------------------------------------------------------------=
-
>Resource Owner                    End User
>Resource Server                    SIP Proxy or SIP Application Server
>Client                                     SIP UA
>Authorization Server             Authorization Server
>
>Do you see a problem with this model?

As I indicated earlier, this is a little different mapping than the one don=
e by 3GPP.

OAuth role                                     IMS role

Resource Owner                           IMS subscriber
Client                                               WWSF
Authorization Server                    WAF
Resource Server                            IMS network

(copied text)

WWSF (WebRTC Web Server Function)

-          Acts as web server, from where the user downloads the IMS access=
 application.
WAF (WebRTC Authorisation Function)

-          Acts as the authorization server (the "facebook"), and issues th=
e authorization tokens.

(The WWSF and WAF can, but does not have to, be located in the same domain =
as the IMS provider.)

I guess the important thing to note here is that the IMS core network (incl=
uding the registrar/S-CSCF and HSS) does not act as Authorization Server, b=
ut rather as the Resource Server.

In fact, the IMS core network itself does not even need to support OAuth2.0=
. Instead the WAF performs mapping and verification between "legacy IMS cre=
dentials" and "web credentials" (e.g. OAuth2.0). The detailed procedures fo=
r that mapping/verification is not specified in the current 3GPP release.

Regards,

Christer





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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D;mso-fareast-language:EN-US=
">Hi,<o:p></o:p></span></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&gt;</span>I am not cl=
ear on the flow of messages in your case.&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&gt;</span>Can you ple=
ase provide a high level diagram that describes that flow?<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">The user contacts the WWS=
F, and uses normal web credentials (e.g. username &#43; password) to reques=
t an access token from the WWSF/WAF, using HTTP. The user then
 sends then includes the access token in the SIP REGISTER request.<o:p></o:=
p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">So, as far as the flow is=
 concerned, I guess the main difference from your draft is that the user do=
es not retrieve the access token using SIP, but using HTTP.<o:p></o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&gt;</span>Also, the t=
oken that is returned by the WAF to the SIP endpoint; is it
<span style=3D"color:#1F497D">&gt;</span>an access token, as per OAuth 2.0,=
 or a user token (ID Token), as per
<span style=3D"color:#1F497D">&gt;</span>OpenID?<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">An access token.<o:p></o:=
p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
</div>
</div>
<div>
<div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">Hi,</span><o:p></o:p></p>
<div>
<div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>What I am proposing is so=
mething along the lines of OpenID, which extends the OAuth to
<span style=3D"color:#1F497D">&gt;</span>authenticate the *user*.<o:p></o:p=
></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>In this case, the mapping=
 between OAuth and SIP would be something like the following:<o:p></o:p></p=
>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>OAuth &nbsp; &nbsp; &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nb=
sp; &nbsp; &nbsp; &nbsp; &nbsp;SIP<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>-------------------------=
--------------------------------------------------<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>Resource Owner &nbsp; &nb=
sp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;End User<o:p></o=
:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>Resource Server &nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;SIP Proxy or SI=
P Application Server<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>Client &nbsp; &nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp; SIP UA<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>Authorization Server &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Authorization Server<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>Do you see a problem with=
 this model?<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">As I indicated earlier, this is a littl=
e different mapping than the one done by 3GPP.</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;=
,&quot;sans-serif&quot;;color:#1F497D">OAuth role&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; IMS role</span></b><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">Resource Owner&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; IMS subscriber</sp=
an><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">Client&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp; WWSF</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">Authorization Server&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp; WAF</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">Resource Server&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; IMS network=
</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">(copied text)</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">WWSF (WebRTC Web Server Function)<o:p></o:p></p>
<p>-<span style=3D"font-size:7.0pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp; </span>Acts as web server, from where the user downloads th=
e IMS access application.<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">WAF (WebRTC Authorisation Function)<o:p></o:p></p>
<p>-<span style=3D"font-size:7.0pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp; </span>Acts as the authorization server (the &#8220;faceboo=
k&#8221;), and issues the authorization tokens.<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">(The WWSF and WAF can, but does not have to, be located in the sam=
e domain as the IMS provider.)<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">I guess the important thing to note here is that the IMS core netw=
ork (including the registrar/S-CSCF and HSS) does not act as Authorization =
Server, but rather as the Resource Server.
<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">In fact, the IMS core network itself does not even need to support=
 OAuth2.0. Instead the WAF performs mapping and verification between &#8220=
;legacy IMS credentials&#8221; and &#8220;web credentials&#8221;
 (e.g. OAuth2.0). The detailed procedures for that mapping/verification is =
not specified in the current 3GPP release.<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#888888">&nbsp;<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#888888">Christer<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><span style=3D"color:#8888=
88"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><span style=3D"color:#8888=
88"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><span style=3D"color:#8888=
88"><o:p></o:p></span></p>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1D4716FEESESSMB209erics_--


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From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>
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Subject: [sipcore] Key-Derivation Authentication Scheme proposal (to replace Digest scheme)
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--089e0158b78efea2f40505117e3a
Content-Type: text/plain; charset=ISO-8859-1

Hi,

During the IETF conference in London and as a response to my presentation
of the update to the SIP Digest, there was interest in the room for
exploring better schemes than the existing Digest scheme.

I have just submitted an draft that describes a new proposal for an
authentication scheme that is based on the PBKDF2 function, to replace the
Digest scheme used by SIP today.

I would really appreciate it if people review it and provide comments.

Regards,
 Rifaat




---------- Forwarded message ----------
From: <internet-drafts@ietf.org>
Date: Fri, Oct 10, 2014 at 9:19 AM
Subject: New Version Notification for
draft-yusef-sipcore-key-derivation-00.txt
To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>



A new version of I-D, draft-yusef-sipcore-key-derivation-00.txt
has been successfully submitted by Rifaat Shekh-Yusef and posted to the
IETF repository.

Name:           draft-yusef-sipcore-key-derivation
Revision:       00
Title:          Key-Derivation Authentication Scheme
Document date:  2014-10-10
Group:          Individual Submission
Pages:          8
URL:
http://www.ietf.org/internet-drafts/draft-yusef-sipcore-key-derivation-00.txt
Status:
https://datatracker.ietf.org/doc/draft-yusef-sipcore-key-derivation/
Htmlized:
http://tools.ietf.org/html/draft-yusef-sipcore-key-derivation-00


Abstract:
   This document defines a Key-Derivation Authentication Scheme, based
   on the PBKDF2 Key Derivation Function (KDF), that could be used with
   the challenge-response authentication framework used by SIP to
   authenticate the user.

   The scheme allows two parties to establish a mutually authenticated
   communication channel based on a shared password, without ever
   sending the password on the wire.





Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

The IETF Secretariat

--089e0158b78efea2f40505117e3a
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>Hi,</div><div><br></div><div>During the IETF conferen=
ce in London and as a response to my presentation of the update to the SIP =
Digest, there was interest in the room for exploring better schemes than th=
e existing Digest scheme.</div><div><br></div><div>I have just submitted an=
 draft that describes a new proposal for an authentication scheme that is b=
ased on the PBKDF2 function, to replace the Digest scheme used by SIP today=
.</div><div><br></div><div>I would really appreciate it if people review it=
 and provide comments.</div><div><br></div><div>Regards,</div><div>=A0Rifaa=
t</div><div><br></div><div><br></div><div><br></div><br><div class=3D"gmail=
_quote">---------- Forwarded message ----------<br>From: <b class=3D"gmail_=
sendername"></b> <span dir=3D"ltr">&lt;<a href=3D"mailto:internet-drafts@ie=
tf.org">internet-drafts@ietf.org</a>&gt;</span><br>Date: Fri, Oct 10, 2014 =
at 9:19 AM<br>Subject: New Version Notification for draft-yusef-sipcore-key=
-derivation-00.txt<br>To: Rifaat Shekh-Yusef &lt;<a href=3D"mailto:rifaat.i=
etf@gmail.com">rifaat.ietf@gmail.com</a>&gt;<br><br><br><br>
A new version of I-D, draft-yusef-sipcore-key-derivation-00.txt<br>
has been successfully submitted by Rifaat Shekh-Yusef and posted to the<br>
IETF repository.<br>
<br>
Name:=A0 =A0 =A0 =A0 =A0 =A0draft-yusef-sipcore-key-derivation<br>
Revision:=A0 =A0 =A0 =A000<br>
Title:=A0 =A0 =A0 =A0 =A0 Key-Derivation Authentication Scheme<br>
Document date:=A0 2014-10-10<br>
Group:=A0 =A0 =A0 =A0 =A0 Individual Submission<br>
Pages:=A0 =A0 =A0 =A0 =A0 8<br>
URL:=A0 =A0 =A0 =A0 =A0 =A0 <a href=3D"http://www.ietf.org/internet-drafts/=
draft-yusef-sipcore-key-derivation-00.txt" target=3D"_blank">http://www.iet=
f.org/internet-drafts/draft-yusef-sipcore-key-derivation-00.txt</a><br>
Status:=A0 =A0 =A0 =A0 =A0<a href=3D"https://datatracker.ietf.org/doc/draft=
-yusef-sipcore-key-derivation/" target=3D"_blank">https://datatracker.ietf.=
org/doc/draft-yusef-sipcore-key-derivation/</a><br>
Htmlized:=A0 =A0 =A0 =A0<a href=3D"http://tools.ietf.org/html/draft-yusef-s=
ipcore-key-derivation-00" target=3D"_blank">http://tools.ietf.org/html/draf=
t-yusef-sipcore-key-derivation-00</a><br>
<br>
<br>
Abstract:<br>
=A0 =A0This document defines a Key-Derivation Authentication Scheme, based<=
br>
=A0 =A0on the PBKDF2 Key Derivation Function (KDF), that could be used with=
<br>
=A0 =A0the challenge-response authentication framework used by SIP to<br>
=A0 =A0authenticate the user.<br>
<br>
=A0 =A0The scheme allows two parties to establish a mutually authenticated<=
br>
=A0 =A0communication channel based on a shared password, without ever<br>
=A0 =A0sending the password on the wire.<br>
<br>
<br>
<br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
The IETF Secretariat<br>
<br>
</div><br></div>

--089e0158b78efea2f40505117e3a--


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References: <10D4F263-37CF-4A74-8F23-EBA9E5010E02@mvryan.org> <CAGL6ep+sc7DKcSF5gXDcGtX5oVMt1-=1w9JMsa1OQJb1AtCrkw@mail.gmail.com> <53FB83FA.8010101@alum.mit.edu> <D020D401.12E169%jon.peterson@neustar.biz> <CAGL6epKSipp6PiHnsC7i9iQKdkss3v0y2CZH4T+zt0Bmh2WRxQ@mail.gmail.com> <D02E41DF.131993%jon.peterson@neustar.biz> <CAGL6epJrS7H+t-CmoW4--MZcsPQJ4em2g2ZiTradLLJEcJEs-A@mail.gmail.com> <D03C75E8.133DCF%jon.peterson@neustar.biz> <CAGL6epJV=00VSba5mUZ2Oi_VpQKQgMA1ddqeRv5aq3wT9fOQbg@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D45BB30@ESESSMB209.ericsson.se> <CAGL6epKjVCEzBFGYMYGv42V7_+Ey5kY+wZhCdqLgqkkMUXmZBg@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D4716FE@ESESSMB209.ericsson.se>
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From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [sipcore] A SIP OAuth use case
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--089e0158b78ef2708e050514c696
Content-Type: text/plain; charset=ISO-8859-1

On Thu, Oct 9, 2014 at 4:38 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
>
>
> >I am not clear on the flow of messages in your case.
>
> >Can you please provide a high level diagram that describes that flow?
>
>
>
> The user contacts the WWSF, and uses normal web credentials (e.g. username
> + password) to request an access token from the WWSF/WAF, using HTTP.
>

This is the part I am still trying to understand. Can you please elaborate
on this?



> The user then sends then includes the access token in the SIP REGISTER
> request.
>
>
>
> So, as far as the flow is concerned, I guess the main difference from your
> draft is that the user does not retrieve the access token using SIP, but
> using HTTP.
>
>
>

In my draft, the only case that the endpoint uses SIP to get a token is
when the endpoint is getting the token from the registrar (using the
Resources Owner Password Credentials grant); otherwise, the endpoint uses
HTTP.



>   >Also, the token that is returned by the WAF to the SIP endpoint; is it
> >an access token, as per OAuth 2.0, or a user token (ID Token), as per >
> OpenID?
>
>
>
> An access token.
>
>
>
> Regards,
>
>
>
> Christer
>
>
>
>  Hi,
>
>
>
> >What I am proposing is something along the lines of OpenID, which
> extends the OAuth to >authenticate the *user*.
>
> >In this case, the mapping between OAuth and SIP would be something like
> the following:
>
> >
>
> >OAuth                                    SIP
>
> >
> ---------------------------------------------------------------------------
>
> >Resource Owner                    End User
>
> >Resource Server                    SIP Proxy or SIP Application Server
>
> >Client                                     SIP UA
>
> >Authorization Server             Authorization Server
>
> >
>
> >Do you see a problem with this model?
>
>
>
> As I indicated earlier, this is a little different mapping than the one
> done by 3GPP.
>
>
>
> *OAuth role                                     IMS role*
>
>
>
> Resource Owner                           IMS subscriber
>
> Client                                               WWSF
>
> Authorization Server                    WAF
>
> Resource Server                            IMS network
>
>
>
> (copied text)
>
>
>
> WWSF (WebRTC Web Server Function)
>
> -          Acts as web server, from where the user downloads the IMS
> access application.
>
> WAF (WebRTC Authorisation Function)
>
> -          Acts as the authorization server (the "facebook"), and issues
> the authorization tokens.
>
>
>
> (The WWSF and WAF can, but does not have to, be located in the same domain
> as the IMS provider.)
>
>
>
> I guess the important thing to note here is that the IMS core network
> (including the registrar/S-CSCF and HSS) does not act as Authorization
> Server, but rather as the Resource Server.
>
>
>
> In fact, the IMS core network itself does not even need to support
> OAuth2.0. Instead the WAF performs mapping and verification between "legacy
> IMS credentials" and "web credentials" (e.g. OAuth2.0). The detailed
> procedures for that mapping/verification is not specified in the current
> 3GPP release.
>
>
>
> Regards,
>
>
>
> Christer
>
>
>
>
>
>
>
>
>

--089e0158b78ef2708e050514c696
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Thu, Oct 9, 2014 at 4:38 PM, Christer Holmberg <span dir=3D"ltr">&lt=
;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christ=
er.holmberg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmai=
l_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left=
:1ex">





<div lang=3D"EN-GB" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Hi,<u></u><u></u></span><=
/p>
<div><span class=3D"">
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d"><u></u>&nbsp;<u></u></=
span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span>I am not cl=
ear on the flow of messages in your case.&nbsp;<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span>Can you ple=
ase provide a high level diagram that describes that flow?<u></u><u></u></p=
>
</div>
</span><div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d"><u></u>&nbsp;<u></u></=
span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">The user contacts the WWS=
F, and uses normal web credentials (e.g. username + password) to request an=
 access token from the WWSF/WAF, using HTTP. </span></p></div></div></div><=
/div></blockquote><div><br></div><div>This is the part I am still trying to=
 understand. Can you please elaborate on this?</div><div><br></div><div>&nb=
sp;</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;borde=
r-left:1px #ccc solid;padding-left:1ex"><div lang=3D"EN-GB" link=3D"blue" v=
link=3D"purple"><div><div><div><p class=3D"MsoNormal"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f=
497d">The user then
 sends then includes the access token in the SIP REGISTER request.<u></u><u=
></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">So, as far as the flow is=
 concerned, I guess the main difference from your draft is that the user do=
es not retrieve the access token using SIP, but using HTTP.<u></u><u></u></=
span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;</span></p><=
/div></div></div></div></blockquote><div><br></div><div>In my draft, the on=
ly case that the endpoint uses SIP to get a token is when the endpoint is g=
etting the token from the registrar (using the Resources Owner Password Cre=
dentials grant); otherwise, the endpoint uses HTTP.</div><div><br></div><di=
v>&nbsp;</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex"><div lang=3D"EN-GB" link=3D"bl=
ue" vlink=3D"purple"><div><div><div><p class=3D"MsoNormal"><span style=3D"f=
ont-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;colo=
r:#1f497d"><u></u></span></p>
</div><span class=3D"">
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span>Also, the t=
oken that is returned by the WAF to the SIP endpoint; is it
<span style=3D"color:#1f497d">&gt;</span>an access token, as per OAuth 2.0,=
 or a user token (ID Token), as per
<span style=3D"color:#1f497d">&gt;</span>OpenID?<u></u><u></u></p>
</div>
</span><div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d"><u></u>&nbsp;<u></u></=
span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">An access token.<u></u><u=
></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Regards,<span class=3D"HO=
EnZb"><font color=3D"#888888"><u></u><u></u></font></span></span></p><span =
class=3D"HOEnZb"><font color=3D"#888888">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Christer<u></u><u></u></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
</font></span></div>
</div><div><div class=3D"h5">
<div>
<div>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Hi,</span><u></u><u></u><=
/p>
<div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&nbsp;</span><u></u><u=
></u></p>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span>What I am p=
roposing is something along the lines of OpenID, which extends the OAuth to
<span style=3D"color:#1f497d">&gt;</span>authenticate the *user*.<u></u><u>=
</u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span>In this cas=
e, the mapping between OAuth and SIP would be something like the following:=
<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span>&nbsp;<u></=
u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span>OAuth &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nb=
sp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;SIP<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span>-----------=
----------------------------------------------------------------<u></u><u><=
/u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span>Resource Ow=
ner &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;En=
d User<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span>Resource Se=
rver &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;S=
IP Proxy or SIP Application Server<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span>Client &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; SIP UA<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span>Authorizati=
on Server &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Authorization Server<u>=
</u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span>&nbsp;<u></=
u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&gt;</span>Do you see =
a problem with this model?<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1f497d">&nbsp;</span><u></u><u=
></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">As I indicated earlier, t=
his is a little different mapping than the one done by 3GPP.</span><u></u><=
u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">&nbsp;</span><u></u><u></=
u></p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:11.0pt;font-family:&quot=
;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">OAuth role&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; IMS role</span></b><=
u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">&nbsp;</span><u></u><u></=
u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Resource Owner&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; IMS =
subscriber</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Client&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; WWSF</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Authorization Server&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; WAF</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Resource Server&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; IMS network</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">&nbsp;</span><u></u><u></=
u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">(copied text)</span><u></=
u><u></u></p>
<p class=3D"MsoNormal">&nbsp;<u></u><u></u></p>
<p class=3D"MsoNormal">WWSF (WebRTC Web Server Function)<u></u><u></u></p>
<p>-<span style=3D"font-size:7.0pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp; </span>Acts as web server, from where the user downloads th=
e IMS access application.<u></u><u></u></p>
<p class=3D"MsoNormal">WAF (WebRTC Authorisation Function)<u></u><u></u></p=
>
<p>-<span style=3D"font-size:7.0pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp; </span>Acts as the authorization server (the &ldquo;faceboo=
k&rdquo;), and issues the authorization tokens.<u></u><u></u></p>
<p class=3D"MsoNormal">&nbsp;<u></u><u></u></p>
<p class=3D"MsoNormal">(The WWSF and WAF can, but does not have to, be loca=
ted in the same domain as the IMS provider.)<u></u><u></u></p>
<p class=3D"MsoNormal">&nbsp;<u></u><u></u></p>
<p class=3D"MsoNormal">I guess the important thing to note here is that the=
 IMS core network (including the registrar/S-CSCF and HSS) does not act as =
Authorization Server, but rather as the Resource Server.
<u></u><u></u></p>
<p class=3D"MsoNormal">&nbsp;<u></u><u></u></p>
<p class=3D"MsoNormal">In fact, the IMS core network itself does not even n=
eed to support OAuth2.0. Instead the WAF performs mapping and verification =
between &ldquo;legacy IMS credentials&rdquo; and &ldquo;web credentials&rdq=
uo;
 (e.g. OAuth2.0). The detailed procedures for that mapping/verification is =
not specified in the current 3GPP release.<u></u><u></u></p>
<p class=3D"MsoNormal">&nbsp;<u></u><u></u></p>
<p class=3D"MsoNormal">Regards,<u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"color:#888888">&nbsp;<u></u><u></u></=
span></p>
<p class=3D"MsoNormal"><span style=3D"color:#888888">Christer<u></u><u></u>=
</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">&nbsp;</span><span style=
=3D"color:#888888"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">&nbsp;</span><span style=
=3D"color:#888888"><u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">&nbsp;</span><span style=
=3D"color:#888888"><u></u><u></u></span></p>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
</div></div></div>
</div>

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

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Thread-Topic: [sipcore] A SIP OAuth use case
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Subject: Re: [sipcore] A SIP OAuth use case
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Hi,

>I am not clear on the flow of messages in your case.
>Can you please provide a high level diagram that describes that flow?

The user contacts the WWSF, and uses normal web credentials (e.g. username =
+ password) to request an access token from the WWSF/WAF, using HTTP.

This is the part I am still trying to understand. Can you please elaborate =
on this?

The user (acting as the OA2 resource owner) downloads the IMS webpage from =
the WWSF (acting as the OA2 client), using HTTP. The WWSF requests an acces=
s token from the WAF (acting as the OA2 authorization server), which mainta=
ins the mapping between the users's web credentials and IMS credentials (ho=
w that mapping is managed is outside the scope of the current 3GPP release)=
.

That access token is then provided to the user, which includes it in the SI=
P REGISTER, in order to get access to the IMS network service (acting as th=
e OA2 resource server).

Regards,

Christer




The user then sends then includes the access token in the SIP REGISTER requ=
est.

So, as far as the flow is concerned, I guess the main difference from your =
draft is that the user does not retrieve the access token using SIP, but us=
ing HTTP.


In my draft, the only case that the endpoint uses SIP to get a token is whe=
n the endpoint is getting the token from the registrar (using the Resources=
 Owner Password Credentials grant); otherwise, the endpoint uses HTTP.


>Also, the token that is returned by the WAF to the SIP endpoint; is it >an=
 access token, as per OAuth 2.0, or a user token (ID Token), as per >OpenID=
?

An access token.

Regards,

Christer

Hi,

>What I am proposing is something along the lines of OpenID, which extends =
the OAuth to >authenticate the *user*.
>In this case, the mapping between OAuth and SIP would be something like th=
e following:
>
>OAuth                                    SIP
>--------------------------------------------------------------------------=
-
>Resource Owner                    End User
>Resource Server                    SIP Proxy or SIP Application Server
>Client                                     SIP UA
>Authorization Server             Authorization Server
>
>Do you see a problem with this model?

As I indicated earlier, this is a little different mapping than the one don=
e by 3GPP.

OAuth role                                     IMS role

Resource Owner                           IMS subscriber
Client                                               WWSF
Authorization Server                    WAF
Resource Server                            IMS network

(copied text)

WWSF (WebRTC Web Server Function)

-          Acts as web server, from where the user downloads the IMS access=
 application.
WAF (WebRTC Authorisation Function)

-          Acts as the authorization server (the "facebook"), and issues th=
e authorization tokens.

(The WWSF and WAF can, but does not have to, be located in the same domain =
as the IMS provider.)

I guess the important thing to note here is that the IMS core network (incl=
uding the registrar/S-CSCF and HSS) does not act as Authorization Server, b=
ut rather as the Resource Server.

In fact, the IMS core network itself does not even need to support OAuth2.0=
. Instead the WAF performs mapping and verification between "legacy IMS cre=
dentials" and "web credentials" (e.g. OAuth2.0). The detailed procedures fo=
r that mapping/verification is not specified in the current 3GPP release.

Regards,

Christer






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</head>
<body lang=3D"EN-GB" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D;mso-fareast-language:EN-US=
">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D;mso-fareast-language:EN-US=
"><o:p>&nbsp;</o:p></span></p>
<div>
<div>
<div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>I am not clear on the flo=
w of messages in your case.&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>Can you please provide a =
high level diagram that describes that flow?<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">The user contacts the WWSF, and uses no=
rmal web credentials (e.g. username &#43; password) to request
 an access token from the WWSF/WAF, using HTTP. </span><o:p></o:p></p>
</div>
</div>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">This is the part I am still trying to understand. Ca=
n you please elaborate on this?<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">The user (acting as the O=
A2 resource owner) downloads the IMS webpage from the WWSF (acting as the O=
A2 client), using HTTP. The WWSF requests an access token
 from the WAF (acting as the OA2 authorization server), which maintains the=
 mapping between the users&#8217;s web credentials and IMS credentials (how=
 that mapping is managed is outside the scope of the current 3GPP release).=
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">That access token is then=
 provided to the user, which includes it in the SIP REGISTER, in order to g=
et access to the IMS network service (acting as the OA2
 resource server).<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">The user then sends then includes the a=
ccess token in the SIP REGISTER request.</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">So, as far as the flow is concerned, I =
guess the main difference from your draft is that the user
 does not retrieve the access token using SIP, but using HTTP.</span><o:p><=
/o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
</div>
</div>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">In my draft, the only case that the endpoint uses SI=
P to get a token is when the endpoint is getting the token from the registr=
ar (using the Resources Owner Password Credentials grant); otherwise, the e=
ndpoint uses HTTP.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>Also, the token that is r=
eturned by the WAF to the SIP endpoint; is it
<span style=3D"color:#1F497D">&gt;</span>an access token, as per OAuth 2.0,=
 or a user token (ID Token), as per
<span style=3D"color:#1F497D">&gt;</span>OpenID?<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">An access token.</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">Regards,</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><span style=3D"color:#8888=
88"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">Christer</span><span style=3D"color:#88=
8888"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><span style=3D"color:#8888=
88"><o:p></o:p></span></p>
</div>
</div>
<div>
<div>
<div>
<div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0c=
m 0cm 0cm 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0cm;margin-=
bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">Hi,</span><o:p></o:p></p>
<div>
<div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>What I am proposing is so=
mething along the lines of OpenID, which extends the OAuth to
<span style=3D"color:#1F497D">&gt;</span>authenticate the *user*.<o:p></o:p=
></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>In this case, the mapping=
 between OAuth and SIP would be something like the following:<o:p></o:p></p=
>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>OAuth &nbsp; &nbsp; &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nb=
sp; &nbsp; &nbsp; &nbsp; &nbsp;SIP<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>-------------------------=
--------------------------------------------------<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>Resource Owner &nbsp; &nb=
sp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;End User<o:p></o=
:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>Resource Server &nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;SIP Proxy or SI=
P Application Server<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>Client &nbsp; &nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp; &nbsp; &nbsp; &nbsp; SIP UA<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>Authorization Server &nbs=
p; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Authorization Server<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&gt;</span>Do you see a problem with=
 this model?<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">As I indicated earlier, this is a littl=
e different mapping than the one done by 3GPP.</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;=
,&quot;sans-serif&quot;;color:#1F497D">OAuth role&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; IMS role</span></b><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">Resource Owner&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; IMS subscriber</sp=
an><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">Client&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp; WWSF</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">Authorization Server&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp; WAF</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">Resource Server&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; IMS network=
</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">(copied text)</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">WWSF (WebRTC Web Server Function)<o:p></o:p></p>
<p>-<span style=3D"font-size:7.0pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp; </span>Acts as web server, from where the user downloads th=
e IMS access application.<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">WAF (WebRTC Authorisation Function)<o:p></o:p></p>
<p>-<span style=3D"font-size:7.0pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp; </span>Acts as the authorization server (the &#8220;faceboo=
k&#8221;), and issues the authorization tokens.<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">(The WWSF and WAF can, but does not have to, be located in the sam=
e domain as the IMS provider.)<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">I guess the important thing to note here is that the IMS core netw=
ork (including the registrar/S-CSCF and HSS) does not act as Authorization =
Server, but rather as the Resource Server.
<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">In fact, the IMS core network itself does not even need to support=
 OAuth2.0. Instead the WAF performs mapping and verification between &#8220=
;legacy IMS credentials&#8221; and &#8220;web credentials&#8221;
 (e.g. OAuth2.0). The detailed procedures for that mapping/verification is =
not specified in the current 3GPP release.<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#888888">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#888888">Christer</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1D475248ESESSMB209erics_--


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Date: Fri, 10 Oct 2014 14:48:34 -0400
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From: worley@ariadne.com (Dale R. Worley)
Sender: worley@ariadne.com (Dale R. Worley)
To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] Key-Derivation Authentication Scheme proposal (to replace Digest scheme)
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> Name:           draft-yusef-sipcore-key-derivation

It's an interesting proposal.  The most important thing is that I'd
like to see a section that describes the motivation, that is, why this
is method better than the current digest scheme.  (Laying out the case
is something that could progress before tidying the technical
presentation.)

There are a bunch of parameters to this algorithm, and it would be
helpful to know what are the considerations that drive the choice of
parameter values.  This is especially true for the iteration-count;
the NIST document proposes a range of 1,000 to 10,000,000, but the
computation power that a typical phone can apply to the problem is
limited.  (Also, is the master-key recomputed just once when the phone
is reconfigured with a new password, or with every message?  For that
matter, how does the phone obtain the salt and iteration count when it
is configured?)

Given the large number of parameters, it would be helpful to have a
small number of particular choices/combinations that are proposed for
use.

I see:

- nonce length
- the hash function used in the KDF
- the salt length
- the iteration count
- the length of the generated master key

Looking at the diagram on page 4, it appears that message 2
(Challenge) contains "pop" which is computed as
"HMAC-SHA256(master-key, digest-string + nonce)".  Message 3
(Response) contains "pop" which is also shown being computed as
"HMAC-SHA256(master-key, digest-string + nonce)", which the server is
supposed to verify, even though it appears that that value was
transmitted in the clear in message 2.  I'm pretty sure that the issue
is that the first "digest-string" is the digest string of message 1
(the original request), whereas the second one is the digest string of
message 3.  But the notation should make that clear.  For that matter,
the notation should make clear which bulleted items are computations,
which are tests, which data items are retrieved from local memory,
which are random numbers constructed at the moment, etc.

Message 4 contains "token", but there is no definition of that.
Indeed, if this scheme is meant only to provide mutual agreement on a
cryptographically stronger key to be input to digest authentication,
there isn't any "token", is there?

Dale


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Subject: Re: [sipcore] Key-Derivation Authentication Scheme proposal (to replace Digest scheme)
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--001a11c25c3edcf3a1050523885b
Content-Type: text/plain; charset=ISO-8859-1

Hi Dale,

Thanks for a thorough review and great feedback, as always.
Please, see my reply inline...

Regards,
 Rifaat



On Fri, Oct 10, 2014 at 2:48 PM, Dale R. Worley <worley@ariadne.com> wrote:

> > Name:           draft-yusef-sipcore-key-derivation
>
> It's an interesting proposal.  The most important thing is that I'd
> like to see a section that describes the motivation, that is, why this
> is method better than the current digest scheme.  (Laying out the case
> is something that could progress before tidying the technical
> presentation.)
>
> Agree. Will add a new section to the next version of the draft.



> There are a bunch of parameters to this algorithm, and it would be
> helpful to know what are the considerations that drive the choice of
> parameter values.  This is especially true for the iteration-count;
> the NIST document proposes a range of 1,000 to 10,000,000, but the
> computation power that a typical phone can apply to the problem is
> limited.


Agree.



> (Also, is the master-key recomputed just once when the phone
> is reconfigured with a new password, or with every message?


Yeah, there are few options here:
I was thinking of creating a new master-key every time the phone refreshes
registrations.
Other options is to tie it to some timer, or the number of times that the
master-key was used.
There is a need for some discussion around this.



> For that
> matter, how does the phone obtain the salt and iteration count when it
> is configured?)
>
>
The phone obtains these from the Challenge request.




> Given the large number of parameters, it would be helpful to have a
> small number of particular choices/combinations that are proposed for
> use.
>
> I see:
>
> - nonce length
> - the hash function used in the KDF
> - the salt length
> - the iteration count
> - the length of the generated master key
>
>
Ok.



> Looking at the diagram on page 4, it appears that message 2
> (Challenge) contains "pop" which is computed as
> "HMAC-SHA256(master-key, digest-string + nonce)".  Message 3
> (Response) contains "pop" which is also shown being computed as
> "HMAC-SHA256(master-key, digest-string + nonce)", which the server is
> supposed to verify, even though it appears that that value was
> transmitted in the clear in message 2.  I'm pretty sure that the issue
> is that the first "digest-string" is the digest string of message 1
> (the original request),


No, this is the pop for the Challenge sent by the server.

The idea is that the server uses the master-key to compute the pop over the
Challenge to prove to the client that it is in possession of the
master-key. When the client receives the Challenge, it will be able to
create the master-key based on the Challenge and then locally compute the
same pop over the Challenge to compare it to the pop received in the
challenge.

The client then creates a pop for the Response to prove to the server that
it is in possession of the password/master-key, and sends it to the server.
When the server receives the response, it will be able to locally compute
the pop of the response and compare it to the pop received from the client.

I will try to clarify this in the next version of the draft.



> whereas the second one is the digest string of
> message 3.  But the notation should make that clear.  For that matter,
> the notation should make clear which bulleted items are computations,
> which are tests, which data items are retrieved from local memory,
> which are random numbers constructed at the moment, etc.
>
>
Agree.




> Message 4 contains "token", but there is no definition of that.
> Indeed, if this scheme is meant only to provide mutual agreement on a
> cryptographically stronger key to be input to digest authentication,
> there isn't any "token", is there?
>
>
Yes, you are right; no tokens here. Copy & paste mistake.


Regards,
 Rifaat




> Dale
>

--001a11c25c3edcf3a1050523885b
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Hi Dale,<div><br></div><div>Thanks for a thorough review a=
nd great feedback, as always.</div><div>Please, see my reply inline...</div=
><div><br></div><div>Regards,</div><div>=A0Rifaat</div><div><br></div><div>=
<br><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, Oct 1=
0, 2014 at 2:48 PM, Dale R. Worley <span dir=3D"ltr">&lt;<a href=3D"mailto:=
worley@ariadne.com" target=3D"_blank">worley@ariadne.com</a>&gt;</span> wro=
te:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;=
border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:=
solid;padding-left:1ex">&gt; Name:=A0 =A0 =A0 =A0 =A0 =A0draft-yusef-sipcor=
e-key-derivation<br>
<br>
It&#39;s an interesting proposal.=A0 The most important thing is that I&#39=
;d<br>
like to see a section that describes the motivation, that is, why this<br>
is method better than the current digest scheme.=A0 (Laying out the case<br=
>
is something that could progress before tidying the technical<br>
presentation.)<br>
<br></blockquote><div>Agree. Will add a new section to the next version of =
the draft.</div><div><br></div><div>=A0</div><blockquote class=3D"gmail_quo=
te" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-col=
or:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
There are a bunch of parameters to this algorithm, and it would be<br>
helpful to know what are the considerations that drive the choice of<br>
parameter values.=A0 This is especially true for the iteration-count;<br>
the NIST document proposes a range of 1,000 to 10,000,000, but the<br>
computation power that a typical phone can apply to the problem is<br>
limited.=A0</blockquote><div><br></div><div>Agree.</div><div><br></div><div=
>=A0<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px=
 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left=
-style:solid;padding-left:1ex"> (Also, is the master-key recomputed just on=
ce when the phone<br>
is reconfigured with a new password, or with every message?=A0 </blockquote=
><div><br></div><div><div>Yeah, there are few options here:</div><div>I was=
 thinking of creating a new master-key every time the phone refreshes regis=
trations.</div><div>Other options is to tie it to some timer, or the number=
 of times that the master-key was used.=A0</div></div><div>There is a need =
for some discussion around this.</div><div><br></div><div>=A0</div><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-wid=
th:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-l=
eft:1ex">For that<br>
matter, how does the phone obtain the salt and iteration count when it<br>
is configured?)<br>
<br></blockquote><div><br></div><div>The phone obtains these from the Chall=
enge request.</div><div><br></div><div><br></div><div>=A0</div><blockquote =
class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1=
px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:=
1ex">
Given the large number of parameters, it would be helpful to have a<br>
small number of particular choices/combinations that are proposed for<br>
use.<br>
<br>
I see:<br>
<br>
- nonce length<br>
- the hash function used in the KDF<br>
- the salt length<br>
- the iteration count<br>
- the length of the generated master key<br>
<br></blockquote><div><br></div><div>Ok.</div><div><br></div><div>=A0</div>=
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
Looking at the diagram on page 4, it appears that message 2<br>
(Challenge) contains &quot;pop&quot; which is computed as<br>
&quot;HMAC-SHA256(master-key, digest-string + nonce)&quot;.=A0 Message 3<br=
>
(Response) contains &quot;pop&quot; which is also shown being computed as<b=
r>
&quot;HMAC-SHA256(master-key, digest-string + nonce)&quot;, which the serve=
r is<br>
supposed to verify, even though it appears that that value was<br>
transmitted in the clear in message 2.=A0 I&#39;m pretty sure that the issu=
e<br>
is that the first &quot;digest-string&quot; is the digest string of message=
 1<br>
(the original request), </blockquote><div><br></div><div>No, this is the po=
p for the Challenge sent by the server.</div><div><br></div><div>The idea i=
s that the server uses the master-key to compute the pop over the Challenge=
 to prove to the client that it is in possession of the master-key. When th=
e client receives the Challenge, it will be able to create the master-key b=
ased on the Challenge and then locally compute the same pop over the Challe=
nge to compare it to the pop received in the challenge.</div><div><br></div=
><div>The client then creates a pop for the Response to prove to the server=
 that it is in possession of the password/master-key, and sends it to the s=
erver. When the server receives the response, it will be able to locally co=
mpute the pop of the response and compare it to the pop received from the c=
lient.</div><div><br></div><div>I will try to clarify this in the next vers=
ion of the draft.</div><div><br></div><div>=A0</div><blockquote class=3D"gm=
ail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-l=
eft-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">wherea=
s the second one is the digest string of<br>
message 3.=A0 But the notation should make that clear.=A0 For that matter,<=
br>
the notation should make clear which bulleted items are computations,<br>
which are tests, which data items are retrieved from local memory,<br>
which are random numbers constructed at the moment, etc.<br>
<br></blockquote><div><br></div><div>Agree.</div><div><br></div><div><br></=
div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px=
 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-=
left-style:solid;padding-left:1ex">
Message 4 contains &quot;token&quot;, but there is no definition of that.<b=
r>
Indeed, if this scheme is meant only to provide mutual agreement on a<br>
cryptographically stronger key to be input to digest authentication,<br>
there isn&#39;t any &quot;token&quot;, is there?<br>
<span class=3D""><font color=3D"#888888"><br></font></span></blockquote><di=
v><br></div><div>Yes, you are right; no tokens here. Copy &amp; paste mista=
ke.</div><div><br></div><div><br></div><div>Regards,</div><div>=A0Rifaat</d=
iv><div><br></div><div><br></div><div>=A0</div><blockquote class=3D"gmail_q=
uote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-c=
olor:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><span class=
=3D""><font color=3D"#888888">
Dale<br>
</font></span></blockquote></div><br></div></div></div>

--001a11c25c3edcf3a1050523885b--


From nobody Mon Oct 13 08:43:53 2014
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Message-ID: <543BF334.5020106@alum.mit.edu>
Date: Mon, 13 Oct 2014 11:43:48 -0400
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] Key-Derivation Authentication Scheme proposal (to replace Digest scheme)
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Rifaat,

Thanks for taking a crack at this. I have a lot of questions.

Is it your plan to use the existing sip authentication headers with a 
new auth-scheme? (IMO that is the preferred path unless there are 
reasons it won't work.) If so, it would be very helpful to show the call 
flows using those terms.

You show an initial-request, then a response, then an confirmation. IIUC 
the challenge will be carried in the sip response to the 
initial-request. Verification of the pop from the server will require 
the digest-string from the initial request - not the response. Is that 
what you want? Then the "response" will be carried in a new sip request, 
that has a different digest-string. I don't think the existing text is 
consistent with this usage.

Note that RFC4474 hasn't been a great success. Part of the trouble is 
that the digest-string includes stuff that often is disturbed in 
existing deployments. STIR is proposing to reduce what is included in 
the digest. That might be the right thing to use here, or it might not 
be enough. Needs consideration.

	Thanks,
	Paul

On 10/10/14 9:26 AM, Rifaat Shekh-Yusef wrote:
> Hi,
>
> During the IETF conference in London and as a response to my
> presentation of the update to the SIP Digest, there was interest in the
> room for exploring better schemes than the existing Digest scheme.
>
> I have just submitted an draft that describes a new proposal for an
> authentication scheme that is based on the PBKDF2 function, to replace
> the Digest scheme used by SIP today.
>
> I would really appreciate it if people review it and provide comments.
>
> Regards,
>   Rifaat
>
>
>
>
> ---------- Forwarded message ----------
> From: ** <internet-drafts@ietf.org <mailto:internet-drafts@ietf.org>>
> Date: Fri, Oct 10, 2014 at 9:19 AM
> Subject: New Version Notification for
> draft-yusef-sipcore-key-derivation-00.txt
> To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com
> <mailto:rifaat.ietf@gmail.com>>
>
>
>
> A new version of I-D, draft-yusef-sipcore-key-derivation-00.txt
> has been successfully submitted by Rifaat Shekh-Yusef and posted to the
> IETF repository.
>
> Name:           draft-yusef-sipcore-key-derivation
> Revision:       00
> Title:          Key-Derivation Authentication Scheme
> Document date:  2014-10-10
> Group:          Individual Submission
> Pages:          8
> URL:
> http://www.ietf.org/internet-drafts/draft-yusef-sipcore-key-derivation-00.txt
> Status: https://datatracker.ietf.org/doc/draft-yusef-sipcore-key-derivation/
> Htmlized: http://tools.ietf.org/html/draft-yusef-sipcore-key-derivation-00
>
>
> Abstract:
>     This document defines a Key-Derivation Authentication Scheme, based
>     on the PBKDF2 Key Derivation Function (KDF), that could be used with
>     the challenge-response authentication framework used by SIP to
>     authenticate the user.
>
>     The scheme allows two parties to establish a mutually authenticated
>     communication channel based on a shared password, without ever
>     sending the password on the wire.
>
>
>
>
>
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org
> <http://tools.ietf.org>.
>
> The IETF Secretariat
>
>
>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>


From nobody Mon Oct 13 10:24:21 2014
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From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] Key-Derivation Authentication Scheme proposal (to replace Digest scheme)
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--089e013c6b9284f3600505512a4d
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Thanks Paul,

Please, see my reply inline...

Regards,
 Rifaat



On Mon, Oct 13, 2014 at 11:43 AM, Paul Kyzivat <pkyzivat@alum.mit.edu>
wrote:

> Rifaat,
>
> Thanks for taking a crack at this. I have a lot of questions.
>
> Is it your plan to use the existing sip authentication headers with a new
> auth-scheme? (IMO that is the preferred path unless there are reasons it
> won't work.)


Definitely Yes.


> If so, it would be very helpful to show the call flows using those terms.
>

Ok.



>
> You show an initial-request, then a response, then an confirmation. IIUC
> the challenge will be carried in the sip response to the initial-request.
> Verification of the pop from the server will require the digest-string from
> the initial request - not the response. Is that what you want?


No. I want the client to validate the pop (the digest-string of the
Challenge) sent by the server in the Challenge to prove to the client it is
in possession of the master-key, and later for the server to validate the
pop (the digest-string of the Response) sent by the client in the Response
to prove to the server that it is in possession of the password/master-key.

I will clarify this point.



> Then the "response" will be carried in a new sip request, that has a
> different digest-string. I don't think the existing text is consistent with
> this usage.
>
> Note that RFC4474 hasn't been a great success. Part of the trouble is that
> the digest-string includes stuff that often is disturbed in existing
> deployments. STIR is proposing to reduce what is included in the digest.
> That might be the right thing to use here, or it might not be enough. Needs
> consideration.
>

Ok. I will take a look at STIR.

Regards,
 Rifaat



>
>         Thanks,
>         Paul
>
> On 10/10/14 9:26 AM, Rifaat Shekh-Yusef wrote:
>
>> Hi,
>>
>> During the IETF conference in London and as a response to my
>> presentation of the update to the SIP Digest, there was interest in the
>> room for exploring better schemes than the existing Digest scheme.
>>
>> I have just submitted an draft that describes a new proposal for an
>> authentication scheme that is based on the PBKDF2 function, to replace
>> the Digest scheme used by SIP today.
>>
>> I would really appreciate it if people review it and provide comments.
>>
>> Regards,
>>   Rifaat
>>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: ** <internet-drafts@ietf.org <mailto:internet-drafts@ietf.org>>
>> Date: Fri, Oct 10, 2014 at 9:19 AM
>> Subject: New Version Notification for
>> draft-yusef-sipcore-key-derivation-00.txt
>> To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com
>> <mailto:rifaat.ietf@gmail.com>>
>>
>>
>>
>> A new version of I-D, draft-yusef-sipcore-key-derivation-00.txt
>> has been successfully submitted by Rifaat Shekh-Yusef and posted to the
>> IETF repository.
>>
>> Name:           draft-yusef-sipcore-key-derivation
>> Revision:       00
>> Title:          Key-Derivation Authentication Scheme
>> Document date:  2014-10-10
>> Group:          Individual Submission
>> Pages:          8
>> URL:
>> http://www.ietf.org/internet-drafts/draft-yusef-sipcore-
>> key-derivation-00.txt
>> Status: https://datatracker.ietf.org/doc/draft-yusef-sipcore-key-
>> derivation/
>> Htmlized: http://tools.ietf.org/html/draft-yusef-sipcore-key-
>> derivation-00
>>
>>
>> Abstract:
>>     This document defines a Key-Derivation Authentication Scheme, based
>>     on the PBKDF2 Key Derivation Function (KDF), that could be used with
>>     the challenge-response authentication framework used by SIP to
>>     authenticate the user.
>>
>>     The scheme allows two parties to establish a mutually authenticated
>>     communication channel based on a shared password, without ever
>>     sending the password on the wire.
>>
>>
>>
>>
>>
>> Please note that it may take a couple of minutes from the time of
>> submission
>> until the htmlized version and diff are available at tools.ietf.org
>> <http://tools.ietf.org>.
>>
>> The IETF Secretariat
>>
>>
>>
>>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
>>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

--089e013c6b9284f3600505512a4d
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Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Thanks Paul,<div><br></div><div>Please, see my reply inlin=
e...</div><div><br></div><div>Regards,</div><div>=A0Rifaat</div><div><br></=
div><div><br></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote=
">On Mon, Oct 13, 2014 at 11:43 AM, Paul Kyzivat <span dir=3D"ltr">&lt;<a h=
ref=3D"mailto:pkyzivat@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit.ed=
u</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margi=
n:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204=
);border-left-style:solid;padding-left:1ex">Rifaat,<br>
<br>
Thanks for taking a crack at this. I have a lot of questions.<br>
<br>
Is it your plan to use the existing sip authentication headers with a new a=
uth-scheme? (IMO that is the preferred path unless there are reasons it won=
&#39;t work.) </blockquote><div><br></div><div>Definitely Yes.</div><div>=
=A0<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px =
0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-=
style:solid;padding-left:1ex">If so, it would be very helpful to show the c=
all flows using those terms.<br></blockquote><div><br></div><div>Ok.</div><=
div><br></div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"marg=
in:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,20=
4);border-left-style:solid;padding-left:1ex">
<br>
You show an initial-request, then a response, then an confirmation. IIUC th=
e challenge will be carried in the sip response to the initial-request. Ver=
ification of the pop from the server will require the digest-string from th=
e initial request - not the response. Is that what you want? </blockquote><=
div><br></div><div>No. I want the client to validate the pop (the digest-st=
ring of the Challenge) sent by the server in the Challenge to prove to the =
client it is in possession of the master-key, and later for the server to v=
alidate the pop=A0(the digest-string of the Response)=A0sent by the client=
=A0in the Response to prove to the server that it is in possession of the p=
assword/master-key.</div><div><br></div><div>I will clarify this point.</di=
v><div><br></div><div>=A0<br></div><blockquote class=3D"gmail_quote" style=
=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(20=
4,204,204);border-left-style:solid;padding-left:1ex">Then the &quot;respons=
e&quot; will be carried in a new sip request, that has a different digest-s=
tring. I don&#39;t think the existing text is consistent with this usage.<b=
r>
<br>
Note that RFC4474 hasn&#39;t been a great success. Part of the trouble is t=
hat the digest-string includes stuff that often is disturbed in existing de=
ployments. STIR is proposing to reduce what is included in the digest. That=
 might be the right thing to use here, or it might not be enough. Needs con=
sideration.<br></blockquote><div><br></div><div>Ok. I will take a look at S=
TIR.</div><div><br></div><div>Regards,</div><div>=A0Rifaat</div><div><br></=
div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px=
 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-=
left-style:solid;padding-left:1ex">
<br>
=A0 =A0 =A0 =A0 Thanks,<br>
=A0 =A0 =A0 =A0 Paul<span><br>
<br>
On 10/10/14 9:26 AM, Rifaat Shekh-Yusef wrote:<br>
</span><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;=
border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:=
solid;padding-left:1ex"><span>
Hi,<br>
<br>
During the IETF conference in London and as a response to my<br>
presentation of the update to the SIP Digest, there was interest in the<br>
room for exploring better schemes than the existing Digest scheme.<br>
<br>
I have just submitted an draft that describes a new proposal for an<br>
authentication scheme that is based on the PBKDF2 function, to replace<br>
the Digest scheme used by SIP today.<br>
<br>
I would really appreciate it if people review it and provide comments.<br>
<br>
Regards,<br>
=A0 Rifaat<br>
<br>
<br>
<br>
<br>
---------- Forwarded message ----------<br></span><span>
From: ** &lt;<a href=3D"mailto:internet-drafts@ietf.org" target=3D"_blank">=
internet-drafts@ietf.org</a> &lt;mailto:<a href=3D"mailto:internet-drafts@i=
etf.org" target=3D"_blank">internet-drafts@ietf.<u></u>org</a>&gt;&gt;<br>
Date: Fri, Oct 10, 2014 at 9:19 AM<br>
Subject: New Version Notification for<br>
draft-yusef-sipcore-key-<u></u>derivation-00.txt<br>
To: Rifaat Shekh-Yusef &lt;<a href=3D"mailto:rifaat.ietf@gmail.com" target=
=3D"_blank">rifaat.ietf@gmail.com</a><br></span><div><div>
&lt;mailto:<a href=3D"mailto:rifaat.ietf@gmail.com" target=3D"_blank">rifaa=
t.ietf@gmail.com</a>&gt;<u></u>&gt;<br>
<br>
<br>
<br>
A new version of I-D, draft-yusef-sipcore-key-<u></u>derivation-00.txt<br>
has been successfully submitted by Rifaat Shekh-Yusef and posted to the<br>
IETF repository.<br>
<br>
Name:=A0 =A0 =A0 =A0 =A0 =A0draft-yusef-sipcore-key-<u></u>derivation<br>
Revision:=A0 =A0 =A0 =A000<br>
Title:=A0 =A0 =A0 =A0 =A0 Key-Derivation Authentication Scheme<br>
Document date:=A0 2014-10-10<br>
Group:=A0 =A0 =A0 =A0 =A0 Individual Submission<br>
Pages:=A0 =A0 =A0 =A0 =A0 8<br>
URL:<br>
<a href=3D"http://www.ietf.org/internet-drafts/draft-yusef-sipcore-key-deri=
vation-00.txt" target=3D"_blank">http://www.ietf.org/internet-<u></u>drafts=
/draft-yusef-sipcore-<u></u>key-derivation-00.txt</a><br>
Status: <a href=3D"https://datatracker.ietf.org/doc/draft-yusef-sipcore-key=
-derivation/" target=3D"_blank">https://datatracker.ietf.org/<u></u>doc/dra=
ft-yusef-sipcore-key-<u></u>derivation/</a><br>
Htmlized: <a href=3D"http://tools.ietf.org/html/draft-yusef-sipcore-key-der=
ivation-00" target=3D"_blank">http://tools.ietf.org/html/<u></u>draft-yusef=
-sipcore-key-<u></u>derivation-00</a><br>
<br>
<br>
Abstract:<br>
=A0 =A0 This document defines a Key-Derivation Authentication Scheme, based=
<br>
=A0 =A0 on the PBKDF2 Key Derivation Function (KDF), that could be used wit=
h<br>
=A0 =A0 the challenge-response authentication framework used by SIP to<br>
=A0 =A0 authenticate the user.<br>
<br>
=A0 =A0 The scheme allows two parties to establish a mutually authenticated=
<br>
=A0 =A0 communication channel based on a shared password, without ever<br>
=A0 =A0 sending the password on the wire.<br>
<br>
<br>
<br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" target=3D"_blank">tools.ietf.org</a><br></div></div>
&lt;<a href=3D"http://tools.ietf.org" target=3D"_blank">http://tools.ietf.o=
rg</a>&gt;.<br>
<br>
The IETF Secretariat<br>
<br>
<br>
<br>
<br>
______________________________<u></u>_________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a><=
br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
>https://www.ietf.org/mailman/<u></u>listinfo/sipcore</a><br>
<br>
</blockquote>
<br>
______________________________<u></u>_________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a><=
br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
>https://www.ietf.org/mailman/<u></u>listinfo/sipcore</a><br>
</blockquote></div><br></div></div>

--089e013c6b9284f3600505512a4d--


From nobody Mon Oct 13 11:15:37 2014
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Date: Mon, 13 Oct 2014 14:15:29 -0400
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] Key-Derivation Authentication Scheme proposal (to replace Digest scheme)
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On 10/13/14 1:24 PM, Rifaat Shekh-Yusef wrote:
> Thanks Paul,
>
> Please, see my reply inline...
>
> Regards,
>   Rifaat
>
>
>
> On Mon, Oct 13, 2014 at 11:43 AM, Paul Kyzivat <pkyzivat@alum.mit.edu
> <mailto:pkyzivat@alum.mit.edu>> wrote:
>
>     Rifaat,
>
>     Thanks for taking a crack at this. I have a lot of questions.
>
>     Is it your plan to use the existing sip authentication headers with
>     a new auth-scheme? (IMO that is the preferred path unless there are
>     reasons it won't work.)
>
>
> Definitely Yes.
>
>     If so, it would be very helpful to show the call flows using those
>     terms.
>
>
> Ok.
>
>
>     You show an initial-request, then a response, then an confirmation.
>     IIUC the challenge will be carried in the sip response to the
>     initial-request. Verification of the pop from the server will
>     require the digest-string from the initial request - not the
>     response. Is that what you want?
>
>
> No. I want the client to validate the pop (the digest-string of the
> Challenge) sent by the server in the Challenge to prove to the client it
> is in possession of the master-key, and later for the server to validate
> the pop (the digest-string of the Response) sent by the client in the
> Response to prove to the server that it is in possession of the
> password/master-key.

Then this will require creating a digest-string over a response. RFC4474 
only defines it for a request, though all the fields it contains are 
also present in a response, so it is straightforward to define.

> I will clarify this point.
>
>
>     Then the "response" will be carried in a new sip request, that has a
>     different digest-string. I don't think the existing text is
>     consistent with this usage.
>
>     Note that RFC4474 hasn't been a great success. Part of the trouble
>     is that the digest-string includes stuff that often is disturbed in
>     existing deployments. STIR is proposing to reduce what is included
>     in the digest. That might be the right thing to use here, or it
>     might not be enough. Needs consideration.
>
>
> Ok. I will take a look at STIR.
>
> Regards,
>   Rifaat
>
>
>              Thanks,
>              Paul
>
>     On 10/10/14 9:26 AM, Rifaat Shekh-Yusef wrote:
>
>         Hi,
>
>         During the IETF conference in London and as a response to my
>         presentation of the update to the SIP Digest, there was interest
>         in the
>         room for exploring better schemes than the existing Digest scheme.
>
>         I have just submitted an draft that describes a new proposal for an
>         authentication scheme that is based on the PBKDF2 function, to
>         replace
>         the Digest scheme used by SIP today.
>
>         I would really appreciate it if people review it and provide
>         comments.
>
>         Regards,
>            Rifaat
>
>
>
>
>         ---------- Forwarded message ----------
>         From: ** <internet-drafts@ietf.org
>         <mailto:internet-drafts@ietf.org>
>         <mailto:internet-drafts@ietf.__org
>         <mailto:internet-drafts@ietf.org>>>
>         Date: Fri, Oct 10, 2014 at 9:19 AM
>         Subject: New Version Notification for
>         draft-yusef-sipcore-key-__derivation-00.txt
>         To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com
>         <mailto:rifaat.ietf@gmail.com>
>         <mailto:rifaat.ietf@gmail.com <mailto:rifaat.ietf@gmail.com>>__>
>
>
>
>         A new version of I-D, draft-yusef-sipcore-key-__derivation-00.txt
>         has been successfully submitted by Rifaat Shekh-Yusef and posted
>         to the
>         IETF repository.
>
>         Name:           draft-yusef-sipcore-key-__derivation
>         Revision:       00
>         Title:          Key-Derivation Authentication Scheme
>         Document date:  2014-10-10
>         Group:          Individual Submission
>         Pages:          8
>         URL:
>         http://www.ietf.org/internet-__drafts/draft-yusef-sipcore-__key-derivation-00.txt
>         <http://www.ietf.org/internet-drafts/draft-yusef-sipcore-key-derivation-00.txt>
>         Status:
>         https://datatracker.ietf.org/__doc/draft-yusef-sipcore-key-__derivation/
>         <https://datatracker.ietf.org/doc/draft-yusef-sipcore-key-derivation/>
>         Htmlized:
>         http://tools.ietf.org/html/__draft-yusef-sipcore-key-__derivation-00
>         <http://tools.ietf.org/html/draft-yusef-sipcore-key-derivation-00>
>
>
>         Abstract:
>              This document defines a Key-Derivation Authentication
>         Scheme, based
>              on the PBKDF2 Key Derivation Function (KDF), that could be
>         used with
>              the challenge-response authentication framework used by SIP to
>              authenticate the user.
>
>              The scheme allows two parties to establish a mutually
>         authenticated
>              communication channel based on a shared password, without ever
>              sending the password on the wire.
>
>
>
>
>
>         Please note that it may take a couple of minutes from the time
>         of submission
>         until the htmlized version and diff are available at
>         tools.ietf.org <http://tools.ietf.org>
>         <http://tools.ietf.org>.
>
>         The IETF Secretariat
>
>
>
>
>         _________________________________________________
>         sipcore mailing list
>         sipcore@ietf.org <mailto:sipcore@ietf.org>
>         https://www.ietf.org/mailman/__listinfo/sipcore
>         <https://www.ietf.org/mailman/listinfo/sipcore>
>
>
>     _________________________________________________
>     sipcore mailing list
>     sipcore@ietf.org <mailto:sipcore@ietf.org>
>     https://www.ietf.org/mailman/__listinfo/sipcore
>     <https://www.ietf.org/mailman/listinfo/sipcore>
>
>


From nobody Mon Oct 13 11:42:20 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: [sipcore] Key-Derivation Authentication Scheme proposal (to replace Digest scheme)
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Subject: Re: [sipcore] Key-Derivation Authentication Scheme proposal (to replace Digest scheme)
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Hi Rifaat,

What do you mean by "uniquely generated" (for the nonce value)?

Regards,

Christer


From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Rifaat Shekh-Y=
usef
Sent: 10 October 2014 16:26
To: sipcore@ietf.org
Subject: [sipcore] Key-Derivation Authentication Scheme proposal (to replac=
e Digest scheme)

Hi,

During the IETF conference in London and as a response to my presentation o=
f the update to the SIP Digest, there was interest in the room for explorin=
g better schemes than the existing Digest scheme.

I have just submitted an draft that describes a new proposal for an authent=
ication scheme that is based on the PBKDF2 function, to replace the Digest =
scheme used by SIP today.

I would really appreciate it if people review it and provide comments.

Regards,
 Rifaat




---------- Forwarded message ----------
From: <internet-drafts@ietf.org<mailto:internet-drafts@ietf.org>>
Date: Fri, Oct 10, 2014 at 9:19 AM
Subject: New Version Notification for draft-yusef-sipcore-key-derivation-00=
.txt
To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com<mailto:rifaat.ietf@gmail.com>=
>



A new version of I-D, draft-yusef-sipcore-key-derivation-00.txt
has been successfully submitted by Rifaat Shekh-Yusef and posted to the
IETF repository.

Name:           draft-yusef-sipcore-key-derivation
Revision:       00
Title:          Key-Derivation Authentication Scheme
Document date:  2014-10-10
Group:          Individual Submission
Pages:          8
URL:            http://www.ietf.org/internet-drafts/draft-yusef-sipcore-key=
-derivation-00.txt
Status:         https://datatracker.ietf.org/doc/draft-yusef-sipcore-key-de=
rivation/
Htmlized:       http://tools.ietf.org/html/draft-yusef-sipcore-key-derivati=
on-00


Abstract:
   This document defines a Key-Derivation Authentication Scheme, based
   on the PBKDF2 Key Derivation Function (KDF), that could be used with
   the challenge-response authentication framework used by SIP to
   authenticate the user.

   The scheme allows two parties to establish a mutually authenticated
   communication channel based on a shared password, without ever
   sending the password on the wire.





Please note that it may take a couple of minutes from the time of submissio=
n
until the htmlized version and diff are available at tools.ietf.org<http://=
tools.ietf.org>.

The IETF Secretariat


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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D;mso-fareast-language:EN-US=
">Hi Rifaat,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D;mso-fareast-language:EN-US=
"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D;mso-fareast-language:EN-US=
">What do you mean by &#8220;uniquely generated&#8221; (for the nonce value=
)?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D;mso-fareast-language:EN-US=
"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D;mso-fareast-language:EN-US=
">Regards,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D;mso-fareast-language:EN-US=
"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D;mso-fareast-language:EN-US=
">Christer<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D;mso-fareast-language:EN-US=
"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><a name=3D"_MailEndCompose"><span style=3D"font-size=
:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497=
D;mso-fareast-language:EN-US"><o:p>&nbsp;</o:p></span></a></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:11.0pt;fo=
nt-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">From:</span></b><span=
 lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&=
quot;sans-serif&quot;"> sipcore [mailto:sipcore-bounces@ietf.org]
<b>On Behalf Of </b>Rifaat Shekh-Yusef<br>
<b>Sent:</b> 10 October 2014 16:26<br>
<b>To:</b> sipcore@ietf.org<br>
<b>Subject:</b> [sipcore] Key-Derivation Authentication Scheme proposal (to=
 replace Digest scheme)<o:p></o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<div>
<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">During the IETF conference in London and as a respon=
se to my presentation of the update to the SIP Digest, there was interest i=
n the room for exploring better schemes than the existing Digest scheme.<o:=
p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">I have just submitted an draft that describes a new =
proposal for an authentication scheme that is based on the PBKDF2 function,=
 to replace the Digest scheme used by SIP today.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">I would really appreciate it if people review it and=
 provide comments.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;Rifaat<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">---------- Forwarded =
message ----------<br>
From: &lt;<a href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.=
org</a>&gt;<br>
Date: Fri, Oct 10, 2014 at 9:19 AM<br>
Subject: New Version Notification for draft-yusef-sipcore-key-derivation-00=
.txt<br>
To: Rifaat Shekh-Yusef &lt;<a href=3D"mailto:rifaat.ietf@gmail.com">rifaat.=
ietf@gmail.com</a>&gt;<br>
<br>
<br>
<br>
A new version of I-D, draft-yusef-sipcore-key-derivation-00.txt<br>
has been successfully submitted by Rifaat Shekh-Yusef and posted to the<br>
IETF repository.<br>
<br>
Name:&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;draft-yusef-sipcore-key-deriv=
ation<br>
Revision:&nbsp; &nbsp; &nbsp; &nbsp;00<br>
Title:&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Key-Derivation Authentication Sche=
me<br>
Document date:&nbsp; 2014-10-10<br>
Group:&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Individual Submission<br>
Pages:&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 8<br>
URL:&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; <a href=3D"http://www.ietf.or=
g/internet-drafts/draft-yusef-sipcore-key-derivation-00.txt" target=3D"_bla=
nk">
http://www.ietf.org/internet-drafts/draft-yusef-sipcore-key-derivation-00.t=
xt</a><br>
Status:&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;<a href=3D"https://datatracker.iet=
f.org/doc/draft-yusef-sipcore-key-derivation/" target=3D"_blank">https://da=
tatracker.ietf.org/doc/draft-yusef-sipcore-key-derivation/</a><br>
Htmlized:&nbsp; &nbsp; &nbsp; &nbsp;<a href=3D"http://tools.ietf.org/html/d=
raft-yusef-sipcore-key-derivation-00" target=3D"_blank">http://tools.ietf.o=
rg/html/draft-yusef-sipcore-key-derivation-00</a><br>
<br>
<br>
Abstract:<br>
&nbsp; &nbsp;This document defines a Key-Derivation Authentication Scheme, =
based<br>
&nbsp; &nbsp;on the PBKDF2 Key Derivation Function (KDF), that could be use=
d with<br>
&nbsp; &nbsp;the challenge-response authentication framework used by SIP to=
<br>
&nbsp; &nbsp;authenticate the user.<br>
<br>
&nbsp; &nbsp;The scheme allows two parties to establish a mutually authenti=
cated<br>
&nbsp; &nbsp;communication channel based on a shared password, without ever=
<br>
&nbsp; &nbsp;sending the password on the wire.<br>
<br>
<br>
<br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" target=3D"_blank">
tools.ietf.org</a>.<br>
<br>
The IETF Secretariat<o:p></o:p></p>
</div>
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From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] Key-Derivation Authentication Scheme proposal (to replace Digest scheme)
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--089e0141a5206baa430505527d97
Content-Type: text/plain; charset=ISO-8859-1

Hi Christer,

It means that the client/server must generate a new unique nonce for each
request/response.
A good example would be a time-stamp that includes micro or even nano
seconds.

I will add more text around this.

Regards,
 Rifaat



On Mon, Oct 13, 2014 at 2:42 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi Rifaat,
>
>
>
> What do you mean by "uniquely generated" (for the nonce value)?
>
>
>
> Regards,
>
>
>
> Christer
>
>
>
>
>
> *From:* sipcore [mailto:sipcore-bounces@ietf.org] *On Behalf Of *Rifaat
> Shekh-Yusef
> *Sent:* 10 October 2014 16:26
> *To:* sipcore@ietf.org
> *Subject:* [sipcore] Key-Derivation Authentication Scheme proposal (to
> replace Digest scheme)
>
>
>
> Hi,
>
>
>
> During the IETF conference in London and as a response to my presentation
> of the update to the SIP Digest, there was interest in the room for
> exploring better schemes than the existing Digest scheme.
>
>
>
> I have just submitted an draft that describes a new proposal for an
> authentication scheme that is based on the PBKDF2 function, to replace the
> Digest scheme used by SIP today.
>
>
>
> I would really appreciate it if people review it and provide comments.
>
>
>
> Regards,
>
>  Rifaat
>
>
>
>
>
>
>
>
>
> ---------- Forwarded message ----------
> From: <internet-drafts@ietf.org>
> Date: Fri, Oct 10, 2014 at 9:19 AM
> Subject: New Version Notification for
> draft-yusef-sipcore-key-derivation-00.txt
> To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
>
>
>
> A new version of I-D, draft-yusef-sipcore-key-derivation-00.txt
> has been successfully submitted by Rifaat Shekh-Yusef and posted to the
> IETF repository.
>
> Name:           draft-yusef-sipcore-key-derivation
> Revision:       00
> Title:          Key-Derivation Authentication Scheme
> Document date:  2014-10-10
> Group:          Individual Submission
> Pages:          8
> URL:
> http://www.ietf.org/internet-drafts/draft-yusef-sipcore-key-derivation-00.txt
> Status:
> https://datatracker.ietf.org/doc/draft-yusef-sipcore-key-derivation/
> Htmlized:
> http://tools.ietf.org/html/draft-yusef-sipcore-key-derivation-00
>
>
> Abstract:
>    This document defines a Key-Derivation Authentication Scheme, based
>    on the PBKDF2 Key Derivation Function (KDF), that could be used with
>    the challenge-response authentication framework used by SIP to
>    authenticate the user.
>
>    The scheme allows two parties to establish a mutually authenticated
>    communication channel based on a shared password, without ever
>    sending the password on the wire.
>
>
>
>
>
> Please note that it may take a couple of minutes from the time of
> submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> The IETF Secretariat
>
>
>

--089e0141a5206baa430505527d97
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Hi Christer,<div><br></div><div>It means that the client/s=
erver must generate a new unique nonce for each request/response.</div><div=
>A good example would be a time-stamp that includes micro or even nano seco=
nds.</div><div><br></div><div>I will add more text around this.</div><div><=
br></div><div>Regards,</div><div>&nbsp;Rifaat<br><div><br></div><div><br></=
div></div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">O=
n Mon, Oct 13, 2014 at 2:42 PM, Christer Holmberg <span dir=3D"ltr">&lt;<a =
href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.h=
olmberg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_qu=
ote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex=
">





<div lang=3D"EN-GB" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Hi Rifaat,<u></u><u></u><=
/span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">What do you mean by &ldqu=
o;uniquely generated&rdquo; (for the nonce value)?<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Regards,<u></u><u></u></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Christer<u></u><u></u></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>&nbsp;<u></u></spa=
n></p>
<p class=3D"MsoNormal"><a name=3D"1490ad15fe208726__MailEndCompose"><span s=
tyle=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&q=
uot;;color:#1f497d"><u></u>&nbsp;<u></u></span></a></p>
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:11.0pt;fo=
nt-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">From:</span></b><span=
 lang=3D"EN-US" style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&=
quot;sans-serif&quot;"> sipcore [mailto:<a href=3D"mailto:sipcore-bounces@i=
etf.org" target=3D"_blank">sipcore-bounces@ietf.org</a>]
<b>On Behalf Of </b>Rifaat Shekh-Yusef<br>
<b>Sent:</b> 10 October 2014 16:26<br>
<b>To:</b> <a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ie=
tf.org</a><br>
<b>Subject:</b> [sipcore] Key-Derivation Authentication Scheme proposal (to=
 replace Digest scheme)<u></u><u></u></span></p><div><div class=3D"h5">
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<div>
<div>
<p class=3D"MsoNormal">Hi,<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">During the IETF conference in London and as a respon=
se to my presentation of the update to the SIP Digest, there was interest i=
n the room for exploring better schemes than the existing Digest scheme.<u>=
</u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I have just submitted an draft that describes a new =
proposal for an authentication scheme that is based on the PBKDF2 function,=
 to replace the Digest scheme used by SIP today.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I would really appreciate it if people review it and=
 provide comments.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Regards,<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;Rifaat<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">---------- Forwarded =
message ----------<br>
From: &lt;<a href=3D"mailto:internet-drafts@ietf.org" target=3D"_blank">int=
ernet-drafts@ietf.org</a>&gt;<br>
Date: Fri, Oct 10, 2014 at 9:19 AM<br>
Subject: New Version Notification for draft-yusef-sipcore-key-derivation-00=
.txt<br>
To: Rifaat Shekh-Yusef &lt;<a href=3D"mailto:rifaat.ietf@gmail.com" target=
=3D"_blank">rifaat.ietf@gmail.com</a>&gt;<br>
<br>
<br>
<br>
A new version of I-D, draft-yusef-sipcore-key-derivation-00.txt<br>
has been successfully submitted by Rifaat Shekh-Yusef and posted to the<br>
IETF repository.<br>
<br>
Name:&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;draft-yusef-sipcore-key-deriv=
ation<br>
Revision:&nbsp; &nbsp; &nbsp; &nbsp;00<br>
Title:&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Key-Derivation Authentication Sche=
me<br>
Document date:&nbsp; 2014-10-10<br>
Group:&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Individual Submission<br>
Pages:&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 8<br>
URL:&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; <a href=3D"http://www.ietf.or=
g/internet-drafts/draft-yusef-sipcore-key-derivation-00.txt" target=3D"_bla=
nk">
http://www.ietf.org/internet-drafts/draft-yusef-sipcore-key-derivation-00.t=
xt</a><br>
Status:&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;<a href=3D"https://datatracker.iet=
f.org/doc/draft-yusef-sipcore-key-derivation/" target=3D"_blank">https://da=
tatracker.ietf.org/doc/draft-yusef-sipcore-key-derivation/</a><br>
Htmlized:&nbsp; &nbsp; &nbsp; &nbsp;<a href=3D"http://tools.ietf.org/html/d=
raft-yusef-sipcore-key-derivation-00" target=3D"_blank">http://tools.ietf.o=
rg/html/draft-yusef-sipcore-key-derivation-00</a><br>
<br>
<br>
Abstract:<br>
&nbsp; &nbsp;This document defines a Key-Derivation Authentication Scheme, =
based<br>
&nbsp; &nbsp;on the PBKDF2 Key Derivation Function (KDF), that could be use=
d with<br>
&nbsp; &nbsp;the challenge-response authentication framework used by SIP to=
<br>
&nbsp; &nbsp;authenticate the user.<br>
<br>
&nbsp; &nbsp;The scheme allows two parties to establish a mutually authenti=
cated<br>
&nbsp; &nbsp;communication channel based on a shared password, without ever=
<br>
&nbsp; &nbsp;sending the password on the wire.<br>
<br>
<br>
<br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" target=3D"_blank">
tools.ietf.org</a>.<br>
<br>
The IETF Secretariat<u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>&nbsp;<u></u></p>
</div>
</div></div></div>
</div>

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

--089e0141a5206baa430505527d97--


From nobody Mon Oct 13 11:59:15 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Thread-Topic: [sipcore] Key-Derivation Authentication Scheme proposal (to replace Digest scheme)
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] Key-Derivation Authentication Scheme proposal (to replace Digest scheme)
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Hi,

A minor editorial comment, related to Paul's request for some flows:=20

I think it would be useful to separate between "SIP response" and "challeng=
e response".

Regards,

Christer

-----Original Message-----
From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Paul Kyzivat
Sent: 13 October 2014 21:15
To: Rifaat Shekh-Yusef
Cc: sipcore@ietf.org
Subject: Re: [sipcore] Key-Derivation Authentication Scheme proposal (to re=
place Digest scheme)

On 10/13/14 1:24 PM, Rifaat Shekh-Yusef wrote:
> Thanks Paul,
>
> Please, see my reply inline...
>
> Regards,
>   Rifaat
>
>
>
> On Mon, Oct 13, 2014 at 11:43 AM, Paul Kyzivat <pkyzivat@alum.mit.edu=20
> <mailto:pkyzivat@alum.mit.edu>> wrote:
>
>     Rifaat,
>
>     Thanks for taking a crack at this. I have a lot of questions.
>
>     Is it your plan to use the existing sip authentication headers with
>     a new auth-scheme? (IMO that is the preferred path unless there are
>     reasons it won't work.)
>
>
> Definitely Yes.
>
>     If so, it would be very helpful to show the call flows using those
>     terms.
>
>
> Ok.
>
>
>     You show an initial-request, then a response, then an confirmation.
>     IIUC the challenge will be carried in the sip response to the
>     initial-request. Verification of the pop from the server will
>     require the digest-string from the initial request - not the
>     response. Is that what you want?
>
>
> No. I want the client to validate the pop (the digest-string of the
> Challenge) sent by the server in the Challenge to prove to the client=20
> it is in possession of the master-key, and later for the server to=20
> validate the pop (the digest-string of the Response) sent by the=20
> client in the Response to prove to the server that it is in possession=20
> of the password/master-key.

Then this will require creating a digest-string over a response. RFC4474 on=
ly defines it for a request, though all the fields it contains are also pre=
sent in a response, so it is straightforward to define.

> I will clarify this point.
>
>
>     Then the "response" will be carried in a new sip request, that has a
>     different digest-string. I don't think the existing text is
>     consistent with this usage.
>
>     Note that RFC4474 hasn't been a great success. Part of the trouble
>     is that the digest-string includes stuff that often is disturbed in
>     existing deployments. STIR is proposing to reduce what is included
>     in the digest. That might be the right thing to use here, or it
>     might not be enough. Needs consideration.
>
>
> Ok. I will take a look at STIR.
>
> Regards,
>   Rifaat
>
>
>              Thanks,
>              Paul
>
>     On 10/10/14 9:26 AM, Rifaat Shekh-Yusef wrote:
>
>         Hi,
>
>         During the IETF conference in London and as a response to my
>         presentation of the update to the SIP Digest, there was interest
>         in the
>         room for exploring better schemes than the existing Digest scheme=
.
>
>         I have just submitted an draft that describes a new proposal for =
an
>         authentication scheme that is based on the PBKDF2 function, to
>         replace
>         the Digest scheme used by SIP today.
>
>         I would really appreciate it if people review it and provide
>         comments.
>
>         Regards,
>            Rifaat
>
>
>
>
>         ---------- Forwarded message ----------
>         From: ** <internet-drafts@ietf.org
>         <mailto:internet-drafts@ietf.org>
>         <mailto:internet-drafts@ietf.__org
>         <mailto:internet-drafts@ietf.org>>>
>         Date: Fri, Oct 10, 2014 at 9:19 AM
>         Subject: New Version Notification for
>         draft-yusef-sipcore-key-__derivation-00.txt
>         To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com
>         <mailto:rifaat.ietf@gmail.com>
>         <mailto:rifaat.ietf@gmail.com=20
> <mailto:rifaat.ietf@gmail.com>>__>
>
>
>
>         A new version of I-D, draft-yusef-sipcore-key-__derivation-00.txt
>         has been successfully submitted by Rifaat Shekh-Yusef and posted
>         to the
>         IETF repository.
>
>         Name:           draft-yusef-sipcore-key-__derivation
>         Revision:       00
>         Title:          Key-Derivation Authentication Scheme
>         Document date:  2014-10-10
>         Group:          Individual Submission
>         Pages:          8
>         URL:
>         http://www.ietf.org/internet-__drafts/draft-yusef-sipcore-__key-d=
erivation-00.txt
>         <http://www.ietf.org/internet-drafts/draft-yusef-sipcore-key-deri=
vation-00.txt>
>         Status:
>         https://datatracker.ietf.org/__doc/draft-yusef-sipcore-key-__deri=
vation/
>         <https://datatracker.ietf.org/doc/draft-yusef-sipcore-key-derivat=
ion/>
>         Htmlized:
>         http://tools.ietf.org/html/__draft-yusef-sipcore-key-__derivation=
-00
>        =20
> <http://tools.ietf.org/html/draft-yusef-sipcore-key-derivation-00>
>
>
>         Abstract:
>              This document defines a Key-Derivation Authentication
>         Scheme, based
>              on the PBKDF2 Key Derivation Function (KDF), that could be
>         used with
>              the challenge-response authentication framework used by SIP =
to
>              authenticate the user.
>
>              The scheme allows two parties to establish a mutually
>         authenticated
>              communication channel based on a shared password, without ev=
er
>              sending the password on the wire.
>
>
>
>
>
>         Please note that it may take a couple of minutes from the time
>         of submission
>         until the htmlized version and diff are available at
>         tools.ietf.org <http://tools.ietf.org>
>         <http://tools.ietf.org>.
>
>         The IETF Secretariat
>
>
>
>
>         _________________________________________________
>         sipcore mailing list
>         sipcore@ietf.org <mailto:sipcore@ietf.org>
>         https://www.ietf.org/mailman/__listinfo/sipcore
>         <https://www.ietf.org/mailman/listinfo/sipcore>
>
>
>     _________________________________________________
>     sipcore mailing list
>     sipcore@ietf.org <mailto:sipcore@ietf.org>
>     https://www.ietf.org/mailman/__listinfo/sipcore
>     <https://www.ietf.org/mailman/listinfo/sipcore>
>
>

_______________________________________________
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] Key-Derivation Authentication Scheme proposal (to replace Digest scheme)
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> From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>

> > Name:           draft-yusef-sipcore-key-derivation

> > (Also, is the master-key recomputed just once when the phone
> > is reconfigured with a new password, or with every message?
> 
> Yeah, there are few options here:
> I was thinking of creating a new master-key every time the phone refreshes
> registrations.
> Other options is to tie it to some timer, or the number of times that the
> master-key was used.
> There is a need for some discussion around this.

As the processing is shown in the draft, the phone doesn't get to
choose when to calculate a master key, it must do so whenever the
parameters presented in the challenge change.

> > For that
> > matter, how does the phone obtain the salt and iteration count when it
> > is configured?)
> >
> The phone obtains these from the Challenge request.

There are a number of inputs to the generic key-derivation function.
This one is presumably configured into the phone by whatever means are
now used for configuring passwords into the phone:
- password

These are presented by the UAS in the 401/407 challenge:
- the identity of the HMAC hash algorithm
- salt
- key-size
- iteration-count

The phone has to do a lot of computation to produce the master key
from these data.  The advantage that is gained is that the master key
is more resistant to a dictionary attack because of the additional
entropy provided by the salt, and the master key is cryptographically
random, providing less opportunity to attack the hash that computes
the message digest.

But at least in this form, the last four data items could vary on a
request-by-request basis, which is probably computationally infeasible
for the phone to deal with.  At the least, you'd want the phone to
cache the results of one master key computation and hope that the
parameters don't change from request to request.

Given that phones are now generally configured by automated means, I'm
wondering if it would not be equally effective to have some process in
the configuration system compute a master key using the NIST method
and then deliver the master key to the phone by the existing
configuration mechanisms.

Dale


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Subject: Re: [sipcore] Key-Derivation Authentication Scheme proposal (to replace Digest scheme)
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On 10/13/14 4:43 PM, Dale R. Worley wrote:

> Given that phones are now generally configured by automated means, I'm
> wondering if it would not be equally effective to have some process in
> the configuration system compute a master key using the NIST method
> and then deliver the master key to the phone by the existing
> configuration mechanisms.

That puts the master key on the wire. To make it safe you would have to 
encrypt it.

	Thanks,
	Paul


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From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
To: "Dale R. Worley" <worley@ariadne.com>
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Subject: Re: [sipcore] Key-Derivation Authentication Scheme proposal (to replace Digest scheme)
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--001a11c341e072113e050561262a
Content-Type: text/plain; charset=ISO-8859-1

Hi Dale,

The master-key is tied to the password selected by the user and it changes
only when the user changes his password.
If we want to be able to change keys on the fly, e.g. during registration
refresh, then a new key must be created that is derived from the
master-key, e.g user-key, which will be used to calculate the pop.
This means that the master-key will only be used to create user-keys, which
will be updated every once in a while.
This will complicate things a bit, but is doable. I will try to capture the
idea in the new version of the draft so we could discuss this.

Regards,
 Rifaat


On Mon, Oct 13, 2014 at 4:43 PM, Dale R. Worley <worley@ariadne.com> wrote:

> > From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
>
> > > Name:           draft-yusef-sipcore-key-derivation
>
> > > (Also, is the master-key recomputed just once when the phone
> > > is reconfigured with a new password, or with every message?
> >
> > Yeah, there are few options here:
> > I was thinking of creating a new master-key every time the phone
> refreshes
> > registrations.
> > Other options is to tie it to some timer, or the number of times that the
> > master-key was used.
> > There is a need for some discussion around this.
>
> As the processing is shown in the draft, the phone doesn't get to
> choose when to calculate a master key, it must do so whenever the
> parameters presented in the challenge change.
>
> > > For that
> > > matter, how does the phone obtain the salt and iteration count when it
> > > is configured?)
> > >
> > The phone obtains these from the Challenge request.
>
> There are a number of inputs to the generic key-derivation function.
> This one is presumably configured into the phone by whatever means are
> now used for configuring passwords into the phone:
> - password
>
> These are presented by the UAS in the 401/407 challenge:
> - the identity of the HMAC hash algorithm
> - salt
> - key-size
> - iteration-count
>
> The phone has to do a lot of computation to produce the master key
> from these data.  The advantage that is gained is that the master key
> is more resistant to a dictionary attack because of the additional
> entropy provided by the salt, and the master key is cryptographically
> random, providing less opportunity to attack the hash that computes
> the message digest.
>
> But at least in this form, the last four data items could vary on a
> request-by-request basis, which is probably computationally infeasible
> for the phone to deal with.  At the least, you'd want the phone to
> cache the results of one master key computation and hope that the
> parameters don't change from request to request.
>
> Given that phones are now generally configured by automated means, I'm
> wondering if it would not be equally effective to have some process in
> the configuration system compute a master key using the NIST method
> and then deliver the master key to the phone by the existing
> configuration mechanisms.
>
> Dale
>

--001a11c341e072113e050561262a
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>Hi Dale,</div><div><br></div><div>The master-key is t=
ied to the password selected by the user and it changes only when the user =
changes his password.=A0</div><div>If we want to be able to change keys on =
the fly, e.g. during registration refresh, then a new key must be created t=
hat is derived from the master-key, e.g user-key, which will be used to cal=
culate the pop.<br></div><div>This means that the master-key will only be u=
sed to create user-keys, which will be updated every once in a while.</div>=
<div>This will complicate things a bit, but is doable. I will try to captur=
e the idea in the new version of the draft so we could discuss this.<br></d=
iv><div><br></div><div>Regards,</div><div>=A0Rifaat</div><div><br></div><di=
v class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Mon, Oct 13, 2014=
 at 4:43 PM, Dale R. Worley <span dir=3D"ltr">&lt;<a href=3D"mailto:worley@=
ariadne.com" target=3D"_blank">worley@ariadne.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">&gt; From: Rifaat Shekh-Yusef &lt;<a href=3D=
"mailto:rifaat.ietf@gmail.com">rifaat.ietf@gmail.com</a>&gt;<br>
<br>
&gt; &gt; Name:=A0 =A0 =A0 =A0 =A0 =A0draft-yusef-sipcore-key-derivation<br=
>
<span class=3D""><br>
&gt; &gt; (Also, is the master-key recomputed just once when the phone<br>
&gt; &gt; is reconfigured with a new password, or with every message?<br>
&gt;<br>
&gt; Yeah, there are few options here:<br>
&gt; I was thinking of creating a new master-key every time the phone refre=
shes<br>
&gt; registrations.<br>
&gt; Other options is to tie it to some timer, or the number of times that =
the<br>
&gt; master-key was used.<br>
&gt; There is a need for some discussion around this.<br>
<br>
</span>As the processing is shown in the draft, the phone doesn&#39;t get t=
o<br>
choose when to calculate a master key, it must do so whenever the<br>
parameters presented in the challenge change.<br>
<span class=3D""><br>
&gt; &gt; For that<br>
&gt; &gt; matter, how does the phone obtain the salt and iteration count wh=
en it<br>
&gt; &gt; is configured?)<br>
&gt; &gt;<br>
&gt; The phone obtains these from the Challenge request.<br>
<br>
</span>There are a number of inputs to the generic key-derivation function.=
<br>
This one is presumably configured into the phone by whatever means are<br>
now used for configuring passwords into the phone:<br>
- password<br>
<br>
These are presented by the UAS in the 401/407 challenge:<br>
- the identity of the HMAC hash algorithm<br>
- salt<br>
- key-size<br>
- iteration-count<br>
<br>
The phone has to do a lot of computation to produce the master key<br>
from these data.=A0 The advantage that is gained is that the master key<br>
is more resistant to a dictionary attack because of the additional<br>
entropy provided by the salt, and the master key is cryptographically<br>
random, providing less opportunity to attack the hash that computes<br>
the message digest.<br>
<br>
But at least in this form, the last four data items could vary on a<br>
request-by-request basis, which is probably computationally infeasible<br>
for the phone to deal with.=A0 At the least, you&#39;d want the phone to<br=
>
cache the results of one master key computation and hope that the<br>
parameters don&#39;t change from request to request.<br>
<br>
Given that phones are now generally configured by automated means, I&#39;m<=
br>
wondering if it would not be equally effective to have some process in<br>
the configuration system compute a master key using the NIST method<br>
and then deliver the master key to the phone by the existing<br>
configuration mechanisms.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Dale<br>
</font></span></blockquote></div><br></div></div>

--001a11c341e072113e050561262a--


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Subject: Re: [sipcore] Key-Derivation Authentication Scheme proposal (to replace Digest scheme)
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> From: Christer Holmberg <christer.holmberg@ericsson.com>

> I think it would be useful to separate between "SIP response" and
> "challenge response".

Yes.  I was losing track of that myself.

Dale


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From: worley@ariadne.com (Dale R. Worley)
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Subject: Re: [sipcore] Key-Derivation Authentication Scheme proposal (to replace Digest scheme)
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> From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>

> It means that the client/server must generate a new unique nonce for each
> request/response.
> A good example would be a time-stamp that includes micro or even nano
> seconds.

I don't think you want to say it that way.  (1) It assumes that the
device knows what time it is.  (2) It assumes that there is only one
message sent during a micro- or nano-second.

As written, a sequence number would suffice.  (You'd need some way of
initializing the sequence number on bootup to ensure you didn't start
counting from zero each time.)  But that raises the question, does the
nonce have to be pseudo-random, or perhaps even cryptographically
random?

My guess is that you want the nonce to be pseudo-random (to prevent
exploitation of regularities in the nonces) but don't need
crypto-random (predictability is not a problem).  So hashing a
sequence number with some sort of per-device identifier would suffice.

Dale


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Subject: Re: [sipcore] Key-Derivation Authentication Scheme proposal (to replace Digest scheme)
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> From: Paul Kyzivat <pkyzivat@alum.mit.edu>
> 
> > Given that phones are now generally configured by automated means, I'm
> > wondering if it would not be equally effective to have some process in
> > the configuration system compute a master key using the NIST method
> > and then deliver the master key to the phone by the existing
> > configuration mechanisms.
> 
> That puts the master key on the wire. To make it safe you would have to 
> encrypt it.

You're stating that a requirement is that the master key does not pass
on the wire in plaintext.

My understanding is that most phones these days (at least in
enterprise settings) obtain their passwords via some sort of
configuration system that runs at boot time -- the password is *not*
entered by the user into the user interface.  And the password passes
over the wire in plaintext.  (Or it might as well be in plaintext, if
we allow the attacker to modify traffic, because the phone has no
prior secret identifier.)

If we really want key secrecy, we'll need more specifications than are
now described in the draft, and they will have to include how the
phone is configured.

Dale


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Subject: Re: [sipcore] Key-Derivation Authentication Scheme proposal (to replace Digest scheme) draft-yusef-sipcore-key-derivation
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> From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>

I think this discussion splits into two parts:

> The master-key is tied to the password selected by the user and it changes
> only when the user changes his password.

As shown in the draft, the master-key depends on the KDF, salt,
key-size, and iteration-count.  All four of those are presented in the
challenge by the UAS.  If the master-key "changes only when the user
changes his password", then there must be a requirement that the UAS
(and every UAS in that authentication realm) must always use the same
values for those parameters (until the password changes).

> If we want to be able to change keys on the fly, e.g. during registration
> refresh, then a new key must be created that is derived from the
> master-key, e.g user-key, which will be used to calculate the pop.
> This means that the master-key will only be used to create user-keys, which
> will be updated every once in a while.
> This will complicate things a bit, but is doable. I will try to capture the
> idea in the new version of the draft so we could discuss this.

The NIST document demands that the master-key is only used to generate
shorter-lived keys, although we don't necessarily have to follow that.

As far as I can tell, the -00 doesn't discuss "changing keys on the
fly".  What are the motivations for that?

Dale


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Hi,

I have just submitted version -01 of the SIP OAuth proposal.
This new version update the first 3 sections, and tries to clarify the
token that is associated with the user, using the OpenID terminology (ID
Token).

Reviews and comments are much appreciated.

Regards,
 Rifaat





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From: <internet-drafts@ietf.org>
Date: Tue, Oct 14, 2014 at 7:44 PM
Subject: New Version Notification for draft-yusef-sipcore-sip-oauth-01.txt
To: Victor Pascual <victor.pascual@quobis.com>, Rifaat Shekh-Yusef <
rifaat.ietf@gmail.com>



A new version of I-D, draft-yusef-sipcore-sip-oauth-01.txt
has been successfully submitted by Rifaat Shekh-Yusef and posted to the
IETF repository.

Name:           draft-yusef-sipcore-sip-oauth
Revision:       01
Title:          The Session Initiation Protocol (SIP) OAuth
Document date:  2014-10-14
Group:          Individual Submission
Pages:          22
URL:
http://www.ietf.org/internet-drafts/draft-yusef-sipcore-sip-oauth-01.txt
Status:
https://datatracker.ietf.org/doc/draft-yusef-sipcore-sip-oauth/
Htmlized:       http://tools.ietf.org/html/draft-yusef-sipcore-sip-oauth-01
Diff:
http://www.ietf.org/rfcdiff?url2=draft-yusef-sipcore-sip-oauth-01

Abstract:
   This document defines an authorization framework for SIP that is
   based on the OAuth 2.0 framework, and adds a simple identity layer on
   top of that, based on the OpenID Connect Core 1.0, to enable Clients
   to verify the identity of the End-User based on the authentication
   performed by an Authorization Server, as well as to obtain basic
   profile information about the End-User.





Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

The IETF Secretariat

--001a113461d21976ab05056aed3c
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<div dir=3D"ltr"><div>Hi,</div><div><br></div><div>I have just submitted ve=
rsion -01 of the SIP OAuth proposal.</div><div>This new version update the =
first 3 sections, and tries to clarify the token that is associated with th=
e user, using the OpenID terminology (ID Token).</div><div><br></div><div>R=
eviews and comments are much appreciated.</div><div><br></div><div>Regards,=
</div><div>=A0Rifaat</div><div><br></div><div><br></div><div><br></div><div=
><br></div><br><div class=3D"gmail_quote">---------- Forwarded message ----=
------<br>From: <b class=3D"gmail_sendername"></b> <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a>&gt;<=
/span><br>Date: Tue, Oct 14, 2014 at 7:44 PM<br>Subject: New Version Notifi=
cation for draft-yusef-sipcore-sip-oauth-01.txt<br>To: Victor Pascual &lt;<=
a href=3D"mailto:victor.pascual@quobis.com">victor.pascual@quobis.com</a>&g=
t;, Rifaat Shekh-Yusef &lt;<a href=3D"mailto:rifaat.ietf@gmail.com">rifaat.=
ietf@gmail.com</a>&gt;<br><br><br><br>
A new version of I-D, draft-yusef-sipcore-sip-oauth-01.txt<br>
has been successfully submitted by Rifaat Shekh-Yusef and posted to the<br>
IETF repository.<br>
<br>
Name:=A0 =A0 =A0 =A0 =A0 =A0draft-yusef-sipcore-sip-oauth<br>
Revision:=A0 =A0 =A0 =A001<br>
Title:=A0 =A0 =A0 =A0 =A0 The Session Initiation Protocol (SIP) OAuth<br>
Document date:=A0 2014-10-14<br>
Group:=A0 =A0 =A0 =A0 =A0 Individual Submission<br>
Pages:=A0 =A0 =A0 =A0 =A0 22<br>
URL:=A0 =A0 =A0 =A0 =A0 =A0 <a href=3D"http://www.ietf.org/internet-drafts/=
draft-yusef-sipcore-sip-oauth-01.txt" target=3D"_blank">http://www.ietf.org=
/internet-drafts/draft-yusef-sipcore-sip-oauth-01.txt</a><br>
Status:=A0 =A0 =A0 =A0 =A0<a href=3D"https://datatracker.ietf.org/doc/draft=
-yusef-sipcore-sip-oauth/" target=3D"_blank">https://datatracker.ietf.org/d=
oc/draft-yusef-sipcore-sip-oauth/</a><br>
Htmlized:=A0 =A0 =A0 =A0<a href=3D"http://tools.ietf.org/html/draft-yusef-s=
ipcore-sip-oauth-01" target=3D"_blank">http://tools.ietf.org/html/draft-yus=
ef-sipcore-sip-oauth-01</a><br>
Diff:=A0 =A0 =A0 =A0 =A0 =A0<a href=3D"http://www.ietf.org/rfcdiff?url2=3Dd=
raft-yusef-sipcore-sip-oauth-01" target=3D"_blank">http://www.ietf.org/rfcd=
iff?url2=3Ddraft-yusef-sipcore-sip-oauth-01</a><br>
<br>
Abstract:<br>
=A0 =A0This document defines an authorization framework for SIP that is<br>
=A0 =A0based on the OAuth 2.0 framework, and adds a simple identity layer o=
n<br>
=A0 =A0top of that, based on the OpenID Connect Core 1.0, to enable Clients=
<br>
=A0 =A0to verify the identity of the End-User based on the authentication<b=
r>
=A0 =A0performed by an Authorization Server, as well as to obtain basic<br>
=A0 =A0profile information about the End-User.<br>
<br>
<br>
<br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
The IETF Secretariat<br>
<br>
</div><br></div>

--001a113461d21976ab05056aed3c--


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From: Richard Shockey <richard@shockey.us>
To: <dispatch@ietf.org>, <sipcore@ietf.org>, <rai@ietf.org>
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As some of you know the SIP Forum and ATIS have been cooperating for the
past year on a series of profile documents for SIP to facilitate all IP
interconnection among service providers.



The Task Force has agreed to make a snapshot of the documents available to
the larger carrier and SIP community to invite informed technical comments.
These are still a =B3work in progress=B2 .



For those of you in the United States this work has taken on a larger
significance since it has the attention of the United States Federal
Communications Commission as was called out in a recent speech by Commissio=
n
Chairman Tom Wheeler.


https://apps.fcc.gov/edocs_public/attachmatch/DOC-329767A1.pdf


 ******




The IP-Network to Network Interface (NNI) Joint Task Force
<http://www.atis.org/PRESS/pressreleases2014/010814.asp> , a cooperative
effort between the ATIS and the SIP Forum, is seeking public comment on two
draft documents:

=20

1.      IP Interconnection Profile.  The IP-NNI Profile Specification
defines a reference architectureand specifications for both the protocol an=
d
media as it appears =B3on-the-wire=B2 at interconnect points.  The
specifications reference commonly used IETF, 3GPP, and other related
industry specifications and identify protocol extensions and capability
information needed for all-IP telephony peering.

2.      IP Interconnection Routing Report.  The IP-NNI Routing Technical
Report documents mechanisms for identifying the preferred IP interconnectio=
n
point for a given TN. This report presents multiple views of current IP
interconnection mechanisms based on aggregate PSTN constructs, interim
solutions based on all-IP routing using a per-TN registry, and a
consideration of hybrid approaches across both mechanisms during the
transition to all-IP.





=20

Members of the IETF SIP Community may access the document directly here.




http://www.sipforum.org/component/option,com_docman/task,cat_view/gid,153/I=
t
emid,261/


Members of ATIS may also Access the documents at:
http://access.atis.org/apps/org/workgroup/ipnni/download.php/18837/latest(f=
o
r ATIS members) or
http://access.atis.org/apps/group_public/document.php?document_id=3D18837&wg_=
a
bbrev=3Dipnni (public access).
=20

Providing Feedback:  Written input, technical feedback only, is requested b=
y
October 29, 2014. However, in light of the short timeframe, input will be
accepted until December 1, 2014.  Submit comments to the Task Force via its
mailing list. To do this, individuals must first be registered as a SIP
Forum Participant Member (Free)



here:http://www.sipforum.org/component/option,com_advanced_registration/tas=
k
,register/.=20



 When completed, subscription to the NNI Task Force mailing list can be
performed at: http://sipforum.org/mailman/listinfo/nni.

=20

For further information on the IP Network to Network Interface (NNI) Joint
Task Force and these above-referenced documents, please contact me or Jim
McEachern, Senior Technical Consultant, ATIS jmceachern@atis.org or me at
the contact information listed below.

=20
=8B=20
Richard Shockey
Shockey Consulting LLC
Chairman of the Board SIP Forum
www.shockey.us
Www.sipforum.org
richard<at>shockey.us
Skype-Linkedin-Facebook rshockey101
PSTN +1 703-593-2683




--B_3496234381_875618
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<html><head><meta http-equiv=3D"content-type" content=3D"text/html; charset=3Dutf=
-8"></head><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -we=
bkit-line-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; fo=
nt-family: Calibri, sans-serif;"><div><h1 class=3D"parseasinTitle " style=3D"fon=
t-size: 14px;"><p class=3D"Body" style=3D"font-family: 'Times New Roman', serif;=
 margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-weight: normal;"><span lang=
=3D"EN-US"><br></span></p><p class=3D"Body" style=3D"font-family: 'Times New Roman=
', serif; margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-weight: normal;"><=
span lang=3D"EN-US">As some of you know the SIP Forum and ATIS have been coope=
rating for the past year on a series of profile documents for SIP to facilit=
ate all IP interconnection among service providers.</span></p><p class=3D"Body=
" style=3D"font-family: 'Times New Roman', serif; margin: 0cm 0cm 0.0001pt; fo=
nt-size: 12pt; font-weight: normal;"><span lang=3D"EN-US"><br></span></p><p cl=
ass=3D"Body" style=3D"font-family: 'Times New Roman', serif; margin: 0cm 0cm 0.0=
001pt; font-size: 12pt; font-weight: normal;"><span lang=3D"EN-US">The Task Fo=
rce has agreed to make a snapshot of the documents available to the larger c=
arrier and SIP community to invite informed technical comments. &nbsp;These =
are still a &#8220;work in progress&#8221; .</span></p><p class=3D"Body" style=
=3D"font-family: 'Times New Roman', serif; margin: 0cm 0cm 0.0001pt; font-size=
: 12pt; font-weight: normal;"><span lang=3D"EN-US"><br></span></p><p class=3D"Bo=
dy" style=3D"font-family: 'Times New Roman', serif; margin: 0cm 0cm 0.0001pt; =
font-size: 12pt; font-weight: normal;">For those of you in the United States=
 this work has taken on a larger significance since it has the attention of =
the United States Federal Communications Commission as was called out in a r=
ecent speech by Commission Chairman Tom Wheeler.</p><p class=3D"Body" style=3D"f=
ont-family: 'Times New Roman', serif; margin: 0cm 0cm 0.0001pt; font-size: 1=
2pt; font-weight: normal;"><br></p><div lang=3D"EN-US" link=3D"#0563C1" vlink=3D"#=
954F72" style=3D"font-family: Helvetica; font-size: 12px;"><div class=3D"WordSec=
tion1" style=3D"page: WordSection1;"><div style=3D"margin: 0in 0in 0.0001pt; fon=
t-size: 11pt; font-family: Calibri, sans-serif;"><a href=3D"https://apps.fcc.g=
ov/edocs_public/attachmatch/DOC-329767A1.pdf" style=3D"color: rgb(149, 79, 114=
);">https://apps.fcc.gov/edocs_public/attachmatch/DOC-329767A1.pdf</a></div>=
<div style=3D"margin: 0in 0in 0.0001pt; font-size: 11pt; font-family: Calibri,=
 sans-serif;"><br></div><div style=3D"margin: 0in 0in 0.0001pt; font-size: 11p=
t; font-family: Calibri, sans-serif;"><br></div><div style=3D"margin: 0in 0in =
0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif;"><o:p>&nbsp;***=
***</o:p></div></div></div><p class=3D"Body" style=3D"font-family: 'Times New Ro=
man', serif; margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-weight: normal;=
"><span lang=3D"EN-US"><br></span></p><p class=3D"Body" style=3D"font-family: 'Tim=
es New Roman', serif; margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-weight=
: normal;"><span lang=3D"EN-US"><br></span></p><p class=3D"Body" style=3D"font-fam=
ily: 'Times New Roman', serif; margin: 0cm 0cm 0.0001pt; font-size: 12pt; fo=
nt-weight: normal;"><span lang=3D"EN-US">The&nbsp;<a href=3D"http://www.atis.org=
/PRESS/pressreleases2014/010814.asp" style=3D"color: purple;">IP-Network to Ne=
twork Interface (NNI) Joint Task Force</a>, a cooperative effort between the=
 ATIS and the SIP Forum, is seeking public comment on two draft documents:<o=
:p></o:p></span></p><p class=3D"Body" style=3D"font-family: 'Times New Roman', s=
erif; margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-weight: normal;"><span=
 lang=3D"EN-US">&nbsp;</span></p><p class=3D"Body" style=3D"font-family: 'Times Ne=
w Roman', serif; margin: 0cm 0cm 0.0001pt 36pt; font-size: 12pt; font-weight=
: normal; text-indent: -18pt;"><span lang=3D"EN-US">1.<span style=3D"font-size: =
7pt; font-family: 'Times New Roman';">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</=
span></span><b><span lang=3D"EN-US">IP Interconnection Profile.&nbsp;&nbsp;</s=
pan></b><span lang=3D"EN-US">The IP-NNI Profile Specification defines a refere=
nce architecture<b></b>and specifications for both the protocol and media as=
 it appears &#8220;on-the-wire&#8221; at interconnect points.&nbsp; The spec=
ifications reference commonly used IETF, 3GPP, and other related industry sp=
ecifications and identify protocol extensions and capability information nee=
ded for all-IP telephony peering.<o:p></o:p></span></p><p class=3D"Body" style=
=3D"font-family: 'Times New Roman', serif; margin: 0cm 0cm 0.0001pt 36pt; font=
-size: 12pt; font-weight: normal; text-indent: -18pt;"><span lang=3D"EN-US">2.=
<span style=3D"font-size: 7pt; font-family: 'Times New Roman';">&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;</span></span><b><span lang=3D"EN-US">IP Interconnection=
 Routing Report</span></b><span lang=3D"EN-US">.&nbsp; The IP-NNI Routing Tech=
nical Report documents mechanisms for identifying the preferred IP interconn=
ection point for a given TN. This report presents multiple views of current =
IP interconnection mechanisms based on aggregate PSTN constructs, interim so=
lutions based on all-IP routing using a per-TN registry, and a consideration=
 of hybrid approaches across both mechanisms during the transition to all-IP=
.&nbsp;<o:p></o:p></span></p><p class=3D"Body" style=3D"font-family: 'Times New =
Roman', serif; margin: 0cm 0cm 0.0001pt 36pt; font-size: 12pt; font-weight: =
normal; text-indent: -18pt;"><span lang=3D"EN-US"><br></span></p><p class=3D"Bod=
y" style=3D"font-family: 'Times New Roman', serif; margin: 0cm 0cm 0.0001pt 36=
pt; font-size: 12pt; font-weight: normal; text-indent: -18pt;"><span lang=3D"E=
N-US"><br></span></p><p class=3D"Body" style=3D"font-family: 'Times New Roman', =
serif; margin: 0cm 0cm 0.0001pt 36pt; font-size: 12pt; font-weight: normal;"=
><b><span lang=3D"EN-US">&nbsp;</span></b></p><p class=3D"MsoNormal" style=3D"marg=
in: 0cm 0cm 0.0001pt; font-size: 11pt; font-weight: normal;"><font face=3D"Tim=
es">Members of the IETF SIP Community may access the document directly here.=
</font></p><p class=3D"MsoNormal" style=3D"margin: 0cm 0cm 0.0001pt; font-size: =
11pt; font-weight: normal;"><font face=3D"Times"><br></font></p><p class=3D"MsoN=
ormal" style=3D"margin: 0cm 0cm 0.0001pt; font-size: 11pt; font-weight: normal=
;"><font face=3D"Times"><br></font></p><p class=3D"MsoNormal" style=3D"margin: 0cm=
 0cm 0.0001pt; font-size: 11pt; font-weight: normal;"><span style=3D"color: rg=
b(0, 0, 255); text-decoration: underline;">http://www.sipforum.org/component=
/option,com_docman/task,cat_view/gid,153/Itemid,261/</span></p><p class=3D"Mso=
Normal" style=3D"margin: 0cm 0cm 0.0001pt; font-size: 11pt; font-weight: norma=
l;"><font face=3D"Times"><br></font></p><p class=3D"MsoNormal" style=3D"font-famil=
y: Calibri, sans-serif; margin: 0cm 0cm 0.0001pt; font-size: 11pt; font-weig=
ht: normal;"><span style=3D"font-size: 12pt; font-family: 'Times New Roman', s=
erif;">Members of ATIS may also Access the documents at:&nbsp;<a href=3D"http:=
//access.atis.org/apps/org/workgroup/ipnni/download.php/18837/latest" style=3D=
"color: purple;">http://access.atis.org/apps/org/workgroup/ipnni/download.ph=
p/18837/latest</a>(for ATIS members) or<o:p></o:p></span></p><p class=3D"MsoNo=
rmal" style=3D"font-family: Calibri, sans-serif; margin: 0cm 0cm 0.0001pt; fon=
t-size: 11pt; font-weight: normal;"><span style=3D"font-size: 12pt; font-famil=
y: 'Times New Roman', serif;"><a href=3D"http://access.atis.org/apps/group_pub=
lic/document.php?document_id=3D18837&amp;wg_abbrev=3Dipnni" style=3D"color: purple=
;">http://access.atis.org/apps/group_public/document.php?document_id=3D18837&a=
mp;wg_abbrev=3Dipnni</a>&nbsp;(public access).<o:p></o:p></span></p><p class=3D"=
Body" style=3D"font-family: 'Times New Roman', serif; margin: 0cm 0cm 0.0001pt=
; font-size: 12pt; font-weight: normal;"><span lang=3D"EN-US" style=3D"font-size=
: 10pt;">&nbsp;</span></p><p class=3D"Body" style=3D"font-family: 'Times New Rom=
an', serif; margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-weight: normal;"=
><b><span lang=3D"EN-US">Providing Feedback</span></b><span lang=3D"EN-US">:&nbs=
p; Written input,&nbsp;<i>technical</i>&nbsp;feedback only, is requested by =
October 29, 2014. However, in light of the short timeframe, input will be ac=
cepted until December 1, 2014.&nbsp; Submit comments to the Task Force via i=
ts mailing list. To do this, individuals must first be registered as a SIP F=
orum Participant Member (Free)&nbsp;</span></p><p class=3D"Body" style=3D"font-f=
amily: 'Times New Roman', serif; margin: 0cm 0cm 0.0001pt; font-size: 12pt; =
font-weight: normal;"><span lang=3D"EN-US"><br></span></p><p class=3D"Body" styl=
e=3D"font-family: 'Times New Roman', serif; margin: 0cm 0cm 0.0001pt; font-siz=
e: 12pt; font-weight: normal;"><span lang=3D"EN-US">here:<a href=3D"http://www.s=
ipforum.org/component/option,com_advanced_registration/task,register/" style=
=3D"color: purple;">http://www.sipforum.org/component/option,com_advanced_regi=
stration/task,register/</a>.&nbsp;</span></p><p class=3D"Body" style=3D"font-fam=
ily: 'Times New Roman', serif; margin: 0cm 0cm 0.0001pt; font-size: 12pt; fo=
nt-weight: normal;"><span lang=3D"EN-US" style=3D"font-size: 12pt; color: rgb(31=
, 73, 125);"><br></span></p><p class=3D"Body" style=3D"font-family: 'Times New R=
oman', serif; margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-weight: normal=
;"><span lang=3D"EN-US" style=3D"font-size: 12pt; color: rgb(31, 73, 125);">&nbs=
p;</span><span lang=3D"EN-US" style=3D"font-size: 12pt;">When completed, subscri=
ption to the NNI Task Force mailing list can be performed at:&nbsp;<a href=3D"=
http://sipforum.org/mailman/listinfo/nni" style=3D"color: purple;">http://sipf=
orum.org/mailman/listinfo/nni</a></span><span class=3D"MsoHyperlink" style=3D"fo=
nt-size: 12pt; color: blue; text-decoration: underline;"><span lang=3D"EN-US">=
.</span></span></p><p class=3D"Body" style=3D"font-family: 'Times New Roman', se=
rif; margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-weight: normal;"><span =
lang=3D"EN-US" style=3D"font-size: 10pt; font-family: Helvetica, sans-serif;">&n=
bsp;</span></p><p class=3D"Body" style=3D"font-family: 'Times New Roman', serif;=
 margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-weight: normal;"><span lang=
=3D"EN-US">For further information on the IP Network to Network Interface (NNI=
) Joint Task Force and these above-referenced documents, please contact me o=
r Jim McEachern, Senior Technical Consultant, ATIS&nbsp;<a href=3D"mailto:jmce=
achern@atis.org" style=3D"color: purple;">jmceachern@atis.org</a>&nbsp;or me a=
t the contact information listed below.<o:p></o:p></span></p><p class=3D"MsoNo=
rmal" style=3D"font-family: Calibri, sans-serif; margin: 0cm 0cm 0.0001pt; fon=
t-size: 11pt; font-weight: normal;"><span style=3D"font-family: Arial, sans-se=
rif;">&nbsp;</span></p></h1><h1 class=3D"parseasinTitle "><font face=3D"Times" s=
tyle=3D"font-size: 16px;">&#8212;&nbsp;</font></h1></div><div><div><font face=3D=
"Times" style=3D"font-size: 16px;">Richard Shockey</font></div><div><font face=
=3D"Times" style=3D"font-size: 16px;">Shockey Consulting LLC</font></div><div><f=
ont face=3D"Times" style=3D"font-size: 16px;">Chairman of the Board SIP Forum</f=
ont></div><div><font face=3D"Times" style=3D"font-size: 16px;">www.shockey.us</f=
ont></div><div><font face=3D"Times" style=3D"font-size: 16px;">Www.sipforum.org<=
/font></div><div><font face=3D"Times" style=3D"font-size: 16px;">richard&lt;at&g=
t;shockey.us</font></div><div><font face=3D"Times" style=3D"font-size: 16px;">Sk=
ype-Linkedin-Facebook rshockey101</font></div><div><font face=3D"Times" style=3D=
"font-size: 16px;">PSTN +1 703-593-2683</font></div><div><br></div></div></b=
ody></html>

--B_3496234381_875618--



From nobody Tue Oct 21 09:23:32 2014
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Date: Tue, 21 Oct 2014 11:23:09 -0500
From: Robert Sparks <rjsparks@nostrum.com>
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Subject: [sipcore] Dealing with the possibility of accepting multiple forked REFER requests
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As Shinji pointed out, the whole reason we have immediate NOTIFY in 
subscription
dialogs is that only one of the 200 OKs to a forked SUBSCRIBE will ever 
make it
back to the subscriber. Trying to carry any information, like the proposed
Refer-Events-At, in the response to a REFER that might fork is going to 
have
the same problem.

A big "Thank You" to everyone who brainstormed possible ways we might do 
things
differently. (I'll summarize and expand on what was discussed before the 
message
is over).

After kicking this around for awhile, I think the right thing to do with 
this
realization is acknowledge the limitation in the mechanism, rather than try
to change the mechanism to avoid it.

My proposal will be to add text to the draft noting that if a REFER 
forks, and
is accepted by more than one recipient, the UA sending the REFER will 
only learn
the Refer-Events-At URI from one of those recipients, and which one is 
out of its
control. So, the extension should only be used when that's ok for the 
application
sending the REFER. That is, either there is enough information to know 
that the
REFER will only be accepted in one place (as is the case if it is sent 
in-dialog,
or Target-Dialog is provided in an out-of-dialog request), or the 
application
doesn't care about tracking the resulting state if the request is accepted
in more than one place. If things would go wrong if the REFER were accepted
in more than one place and the referrer didn't learn about each of them, 
then
the application must not use this extension (falling back into a 
6665-subscription
creating REFER instead).

I know that as engineers, we find this kind of shortcoming distateful.
But, pragmatically, there isn't a strong demand for solving the general 
problem.

If I'm wrong about that, and we decide we must figure out how to get 
information
back from every place a forked REFER is accepted, I see three basic paths
to choose from (based on the recent conversations, and some really old 
ones -
remember HERFP - this is like that, only harder). Again, this is talking 
only
about cases where the REFER might fork - we'll deal with in-dialog or other
non-forking cases separately.

1) Modify proxy behavior to aggregate the responses. As pointed out on 
the list
already, this really a non-starter - it means the proxy will have to hold
the first 200 in case others arrive, forcing it into the bad places of long
non-invite transactions as detailed in RFC4320 and RFC4321.

2) Send an out-of-dialog request from each accepting endpoint back in the
other direction.  We'd have to work through making sure the request would
get back to the right place (probably using GRUUs). We'd have to choose
or define a method (NOTIFY has been suggested, and the "must be in a dialog"
problems with that pointed out. INFO has the same problem. We could abuse
MESSAGE, but it's probably better just to define a new method if we decide
to go down this path.

3) Solve the problem with dialogs. Probably by starting with a new INVITE
that negotiates no (or baseline held) media.  Wherever that INVITE is
accepted, you end up with separate dialogs.  A separate REFER could then
be sent within each dialog, and would be guaranteed to be accepted in only
one place. This would mean defining behavior for that INVITE (probably
involving a Require extension), and detailing when that INVITE dialog
would shut down.

Does anyone see any other class of alternatives?

Again, my recommendation is that we don't pursue _any_ of them, and
go with the recognize-and-restrict approach I suggest at the beginning
of this message.

RjS


From nobody Tue Oct 21 09:57:17 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Robert Sparks <rjsparks@nostrum.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: [sipcore] Dealing with the possibility of accepting multiple forked REFER requests
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Subject: Re: [sipcore] Dealing with the possibility of accepting multiple forked REFER requests
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Hi,

+1 for "don't pursue _any_ of them".

Nothing prevents people from doing 3), but it's nothing we need to describe=
 or mandate. I think it would be restrictive, as there is no guarantee that=
 a REFER and INVITE would get forked to the same destinations, and we would=
 never reach a destination that supports REFER, but does not support INVITE=
.

Regards,

Christer=20

-----Original Message-----
From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Robert Sparks
Sent: 21 October 2014 19:23
To: sipcore@ietf.org
Subject: [sipcore] Dealing with the possibility of accepting multiple forke=
d REFER requests

As Shinji pointed out, the whole reason we have immediate NOTIFY in subscri=
ption dialogs is that only one of the 200 OKs to a forked SUBSCRIBE will ev=
er make it back to the subscriber. Trying to carry any information, like th=
e proposed Refer-Events-At, in the response to a REFER that might fork is g=
oing to have the same problem.

A big "Thank You" to everyone who brainstormed possible ways we might do th=
ings differently. (I'll summarize and expand on what was discussed before t=
he message is over).

After kicking this around for awhile, I think the right thing to do with th=
is realization is acknowledge the limitation in the mechanism, rather than =
try to change the mechanism to avoid it.

My proposal will be to add text to the draft noting that if a REFER forks, =
and is accepted by more than one recipient, the UA sending the REFER will o=
nly learn the Refer-Events-At URI from one of those recipients, and which o=
ne is out of its control. So, the extension should only be used when that's=
 ok for the application sending the REFER. That is, either there is enough =
information to know that the REFER will only be accepted in one place (as i=
s the case if it is sent in-dialog, or Target-Dialog is provided in an out-=
of-dialog request), or the application doesn't care about tracking the resu=
lting state if the request is accepted in more than one place. If things wo=
uld go wrong if the REFER were accepted in more than one place and the refe=
rrer didn't learn about each of them, then the application must not use thi=
s extension (falling back into a 6665-subscription creating REFER instead).

I know that as engineers, we find this kind of shortcoming distateful.
But, pragmatically, there isn't a strong demand for solving the general pro=
blem.

If I'm wrong about that, and we decide we must figure out how to get inform=
ation back from every place a forked REFER is accepted, I see three basic p=
aths to choose from (based on the recent conversations, and some really old=
 ones - remember HERFP - this is like that, only harder). Again, this is ta=
lking only about cases where the REFER might fork - we'll deal with in-dial=
og or other non-forking cases separately.

1) Modify proxy behavior to aggregate the responses. As pointed out on the =
list already, this really a non-starter - it means the proxy will have to h=
old the first 200 in case others arrive, forcing it into the bad places of =
long non-invite transactions as detailed in RFC4320 and RFC4321.

2) Send an out-of-dialog request from each accepting endpoint back in the o=
ther direction.  We'd have to work through making sure the request would ge=
t back to the right place (probably using GRUUs). We'd have to choose or de=
fine a method (NOTIFY has been suggested, and the "must be in a dialog"
problems with that pointed out. INFO has the same problem. We could abuse M=
ESSAGE, but it's probably better just to define a new method if we decide t=
o go down this path.

3) Solve the problem with dialogs. Probably by starting with a new INVITE t=
hat negotiates no (or baseline held) media.  Wherever that INVITE is accept=
ed, you end up with separate dialogs.  A separate REFER could then be sent =
within each dialog, and would be guaranteed to be accepted in only one plac=
e. This would mean defining behavior for that INVITE (probably involving a =
Require extension), and detailing when that INVITE dialog would shut down.

Does anyone see any other class of alternatives?

Again, my recommendation is that we don't pursue _any_ of them, and go with=
 the recognize-and-restrict approach I suggest at the beginning of this mes=
sage.

RjS

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


From nobody Tue Oct 21 10:00:19 2014
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To: OKUMURA Shinji <ietf.shinji@gmail.com>, sipcore@ietf.org
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Subject: Re: [sipcore] I-D Action: draft-sparks-sipcore-refer-clarifications-04.txt
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[as participant]

On 8/19/14 07:13, OKUMURA Shinji wrote:
> If the 6665 compliant UAS supports REFER, then
>   1) In a dialog created by INVITE, a local Contact of the UAS MUST be GRUU.

This is actually a clarification for RFC 6665, not REFER. I still owe 
the WG a draft clarifying exactly this point, and apologize for being 
rather deeply behind on coming up with this.

>   2) If the UAS supports a scenario for a transfer, then it SHOULD support tdialog.

This is a good point; I believe it should be couched in terms of 
pointing out that both referers and referees compliant with 6665 are 
bound by section 6.5 of RFC 6665, and that the appropriate mechanism for 
REFER is RFC 4538. (RFC 6665 implies that it expects event packages to 
call this out explicitly, so I think that including it in the 
clarifications document would be good form)

>   3) When the UAS receives REFER within the dialog, if referer supports tdialog,
>      then the UAS MAY respond by 3xx "Extension Required" with "Required: tdialog" and "Contact: GRUUofReferee",
>      else the UAS MAY accept the REFER.

It looks like you're trying to add behavior to RFC6665 referees to force 
pre-RFC6665 referers that support tdialog (but have elected not to use 
it) to use it. I'm not sure this adds much value -- the prospect that a 
client would know how to do this but decide not to seem kind of slim -- 
but I don't really see any harm.

>   4) When the UAS receives REFER out of the dialog, UAS MAY accept the REFER.

I'm not sure this needs to be said. I think it's reasonable to assume 
that it would accept the REFER, subject to its local policy. Adding 
words to reiterate obvious behavior makes the document longer, and 
therefore worse.

>   1) If a UAS's local Contact inside an existing dialog is gruu, then the 6665 compliant UAC MUST use REFER out of dialog.

There's been a lot of discussion around this wording, and the desire 
seems to be to leave the door open for the explicit subscription 
mechanism that Robert is also working on. The resulting text is the 
first sentence of section 4. It basically says the same thing as you're 
proposing, except that it doesn't need to be updated by the explicit 
subscription mechanism.

>   2) If the UAC supports a scenario for a transfer, then it SHOULD support tdialog.

This is covered by (2), above.

>   3) If a UAS does not support both of gruu and tdialog, then the 6665 compliant UAC MAY use REFER within dialog.

I think this is covered adequately in the final paragraph of section 4.

>   4) If a UAC received 3xx "Extension Required" response for REFER that contains "Required: tdialog",
>      then UAS MAY send REFER out of dialog. The REFER request SHOULD contain "Targer-Dialog" header.

This seems to be trying to impose requirements on legacy, already 
deployed clients. I'm not sure how it provides any value without the use 
of a time machine.

/a


From nobody Tue Oct 21 10:04:44 2014
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On 10/21/14 11:23, Robert Sparks wrote:
> I know that as engineers, we find this kind of shortcoming distateful.
> But, pragmatically, there isn't a strong demand for solving the 
> general problem.

Agreed on both counts.

> Again, my recommendation is that we don't pursue _any_ of them, and
> go with the recognize-and-restrict approach I suggest at the beginning
> of this message.

I agree with this approach. All three proposals are more complicated 
than the root problem we're trying to solve (unwanted NOTIFY messages); 
and, in fact, (2) is basically a proposal to send NOTIFY messages as 
part of an attempt to not send NOTIFY messages.

So, to be clear: I agree that we should document and accept the 
limitation imposed by forking in this scenario, but not add any 
mechanism to mitigate it.

/a


From nobody Tue Oct 21 10:20:55 2014
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[as individual]

On 10/21/14 12:23 PM, Robert Sparks wrote:

> Again, my recommendation is that we don't pursue _any_ of them, and
> go with the recognize-and-restrict approach I suggest at the beginning
> of this message.

I agree.

I'll also mention that one of the reasons people want this solution is 
that they want a solution that doesn't require the use of GRUUs. So 
solutions that themselves require GRUU are non-starters.

	Thanks,
	Paul


From nobody Tue Oct 21 11:12:05 2014
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I'd like some clarity on what "must use a gruu" means.
Does this mean the contact must have a "gr" parameter, or only that it 
must be globally routable? (Note that 3261 has always required contact 
URIs to be globally routable.) The key difference is that with a "gr" 
parameter the other end can verify that the uri has been asserted to be 
globally routable.

	Thanks,
	Paul

On 10/21/14 1:00 PM, Adam Roach wrote:
> [as participant]
>
> On 8/19/14 07:13, OKUMURA Shinji wrote:
>> If the 6665 compliant UAS supports REFER, then
>>   1) In a dialog created by INVITE, a local Contact of the UAS MUST be
>> GRUU.
>
> This is actually a clarification for RFC 6665, not REFER. I still owe
> the WG a draft clarifying exactly this point, and apologize for being
> rather deeply behind on coming up with this.
>
>>   2) If the UAS supports a scenario for a transfer, then it SHOULD
>> support tdialog.
>
> This is a good point; I believe it should be couched in terms of
> pointing out that both referers and referees compliant with 6665 are
> bound by section 6.5 of RFC 6665, and that the appropriate mechanism for
> REFER is RFC 4538. (RFC 6665 implies that it expects event packages to
> call this out explicitly, so I think that including it in the
> clarifications document would be good form)
>
>>   3) When the UAS receives REFER within the dialog, if referer
>> supports tdialog,
>>      then the UAS MAY respond by 3xx "Extension Required" with
>> "Required: tdialog" and "Contact: GRUUofReferee",
>>      else the UAS MAY accept the REFER.
>
> It looks like you're trying to add behavior to RFC6665 referees to force
> pre-RFC6665 referers that support tdialog (but have elected not to use
> it) to use it. I'm not sure this adds much value -- the prospect that a
> client would know how to do this but decide not to seem kind of slim --
> but I don't really see any harm.
>
>>   4) When the UAS receives REFER out of the dialog, UAS MAY accept the
>> REFER.
>
> I'm not sure this needs to be said. I think it's reasonable to assume
> that it would accept the REFER, subject to its local policy. Adding
> words to reiterate obvious behavior makes the document longer, and
> therefore worse.
>
>>   1) If a UAS's local Contact inside an existing dialog is gruu, then
>> the 6665 compliant UAC MUST use REFER out of dialog.
>
> There's been a lot of discussion around this wording, and the desire
> seems to be to leave the door open for the explicit subscription
> mechanism that Robert is also working on. The resulting text is the
> first sentence of section 4. It basically says the same thing as you're
> proposing, except that it doesn't need to be updated by the explicit
> subscription mechanism.
>
>>   2) If the UAC supports a scenario for a transfer, then it SHOULD
>> support tdialog.
>
> This is covered by (2), above.
>
>>   3) If a UAS does not support both of gruu and tdialog, then the 6665
>> compliant UAC MAY use REFER within dialog.
>
> I think this is covered adequately in the final paragraph of section 4.
>
>>   4) If a UAC received 3xx "Extension Required" response for REFER
>> that contains "Required: tdialog",
>>      then UAS MAY send REFER out of dialog. The REFER request SHOULD
>> contain "Targer-Dialog" header.
>
> This seems to be trying to impose requirements on legacy, already
> deployed clients. I'm not sure how it provides any value without the use
> of a time machine.
>
> /a
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>


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fyi


-------- Forwarded Message --------
Subject: 	New Version Notification for 
draft-sparks-sipcore-refer-explicit-subscription-02.txt
Date: 	Tue, 21 Oct 2014 12:26:36 -0700
From: 	internet-drafts@ietf.org
To: 	Robert Sparks <rjsparks@nostrum.com>, Robert Sparks 
<rjsparks@nostrum.com>



A new version of I-D, draft-sparks-sipcore-refer-explicit-subscription-02.txt
has been successfully submitted by Robert Sparks and posted to the
IETF repository.

Name:		draft-sparks-sipcore-refer-explicit-subscription
Revision:	02
Title:		Explicit Subscriptions for the REFER Method
Document date:	2014-10-21
Group:		Individual Submission
Pages:		13
URL:            http://www.ietf.org/internet-drafts/draft-sparks-sipcore-refer-explicit-subscription-02.txt
Status:         https://datatracker.ietf.org/doc/draft-sparks-sipcore-refer-explicit-subscription/
Htmlized:       http://tools.ietf.org/html/draft-sparks-sipcore-refer-explicit-subscription-02
Diff:           http://www.ietf.org/rfcdiff?url2=draft-sparks-sipcore-refer-explicit-subscription-02

Abstract:
    The Session Initiation Protocol (SIP) REFER request, as defined by
    RFC3515, triggers an implicit SIP-Specific Event Notification
    framework subscription.  Conflating the start of the subscription
    with handling the REFER request makes negotiating SUBSCRIBE
    extensions impossible, and complicates avoiding SIP dialog sharing.
    This document defines extensions to REFER to remove the implicit
    subscription and, if desired, replace it with an explicit one.

                                                                                   


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

The IETF Secretariat




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            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject:
            </th>
            <td>New Version Notification for
              draft-sparks-sipcore-refer-explicit-subscription-02.txt</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
            <td>Tue, 21 Oct 2014 12:26:36 -0700</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
            <td><a class="moz-txt-link-abbreviated" href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a></td>
          </tr>
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            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
            <td>Robert Sparks <a class="moz-txt-link-rfc2396E" href="mailto:rjsparks@nostrum.com">&lt;rjsparks@nostrum.com&gt;</a>, Robert
              Sparks <a class="moz-txt-link-rfc2396E" href="mailto:rjsparks@nostrum.com">&lt;rjsparks@nostrum.com&gt;</a></td>
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      <pre>A new version of I-D, draft-sparks-sipcore-refer-explicit-subscription-02.txt
has been successfully submitted by Robert Sparks and posted to the
IETF repository.

Name:		draft-sparks-sipcore-refer-explicit-subscription
Revision:	02
Title:		Explicit Subscriptions for the REFER Method
Document date:	2014-10-21
Group:		Individual Submission
Pages:		13
URL:            <a class="moz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-sparks-sipcore-refer-explicit-subscription-02.txt">http://www.ietf.org/internet-drafts/draft-sparks-sipcore-refer-explicit-subscription-02.txt</a>
Status:         <a class="moz-txt-link-freetext" href="https://datatracker.ietf.org/doc/draft-sparks-sipcore-refer-explicit-subscription/">https://datatracker.ietf.org/doc/draft-sparks-sipcore-refer-explicit-subscription/</a>
Htmlized:       <a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-sparks-sipcore-refer-explicit-subscription-02">http://tools.ietf.org/html/draft-sparks-sipcore-refer-explicit-subscription-02</a>
Diff:           <a class="moz-txt-link-freetext" href="http://www.ietf.org/rfcdiff?url2=draft-sparks-sipcore-refer-explicit-subscription-02">http://www.ietf.org/rfcdiff?url2=draft-sparks-sipcore-refer-explicit-subscription-02</a>

Abstract:
   The Session Initiation Protocol (SIP) REFER request, as defined by
   RFC3515, triggers an implicit SIP-Specific Event Notification
   framework subscription.  Conflating the start of the subscription
   with handling the REFER request makes negotiating SUBSCRIBE
   extensions impossible, and complicates avoiding SIP dialog sharing.
   This document defines extensions to REFER to remove the implicit
   subscription and, if desired, replace it with an explicit one.

                                                                                  


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

The IETF Secretariat

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From nobody Thu Oct 23 01:31:35 2014
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From: OKUMURA Shinji <ietf.shinji@gmail.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 23 Oct 2014 17:31:19 +0900
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Subject: Re: [sipcore] Dealing with the possibility of accepting multiple forked REFER requests
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Hi,

thank you for an attentive summary.
I agree your recognize-and-restrict approach.

Shinji

>As Shinji pointed out, the whole reason we have immediate NOTIFY in 
>subscription
>dialogs is that only one of the 200 OKs to a forked SUBSCRIBE will ever 
>make it
>back to the subscriber. Trying to carry any information, like the proposed
>Refer-Events-At, in the response to a REFER that might fork is going to 
>have
>the same problem.
>
>A big "Thank You" to everyone who brainstormed possible ways we might do 
>things
>differently. (I'll summarize and expand on what was discussed before the 
>message
>is over).
>
>After kicking this around for awhile, I think the right thing to do with 
>this
>realization is acknowledge the limitation in the mechanism, rather than try
>to change the mechanism to avoid it.
>
>My proposal will be to add text to the draft noting that if a REFER 
>forks, and
>is accepted by more than one recipient, the UA sending the REFER will 
>only learn
>the Refer-Events-At URI from one of those recipients, and which one is 
>out of its
>control. So, the extension should only be used when that's ok for the 
>application
>sending the REFER. That is, either there is enough information to know 
>that the
>REFER will only be accepted in one place (as is the case if it is sent 
>in-dialog,
>or Target-Dialog is provided in an out-of-dialog request), or the 
>application
>doesn't care about tracking the resulting state if the request is accepted
>in more than one place. If things would go wrong if the REFER were accepted
>in more than one place and the referrer didn't learn about each of them, 
>then
>the application must not use this extension (falling back into a 
>6665-subscription
>creating REFER instead).
>
>I know that as engineers, we find this kind of shortcoming distateful.
>But, pragmatically, there isn't a strong demand for solving the general 
>problem.
>
>If I'm wrong about that, and we decide we must figure out how to get 
>information
>back from every place a forked REFER is accepted, I see three basic paths
>to choose from (based on the recent conversations, and some really old 
>ones -
>remember HERFP - this is like that, only harder). Again, this is talking 
>only
>about cases where the REFER might fork - we'll deal with in-dialog or other
>non-forking cases separately.
>
>1) Modify proxy behavior to aggregate the responses. As pointed out on 
>the list
>already, this really a non-starter - it means the proxy will have to hold
>the first 200 in case others arrive, forcing it into the bad places of long
>non-invite transactions as detailed in RFC4320 and RFC4321.
>
>2) Send an out-of-dialog request from each accepting endpoint back in the
>other direction.  We'd have to work through making sure the request would
>get back to the right place (probably using GRUUs). We'd have to choose
>or define a method (NOTIFY has been suggested, and the "must be in a dialog"
>problems with that pointed out. INFO has the same problem. We could abuse
>MESSAGE, but it's probably better just to define a new method if we decide
>to go down this path.
>
>3) Solve the problem with dialogs. Probably by starting with a new INVITE
>that negotiates no (or baseline held) media.  Wherever that INVITE is
>accepted, you end up with separate dialogs.  A separate REFER could then
>be sent within each dialog, and would be guaranteed to be accepted in only
>one place. This would mean defining behavior for that INVITE (probably
>involving a Require extension), and detailing when that INVITE dialog
>would shut down.
>
>Does anyone see any other class of alternatives?
>
>Again, my recommendation is that we don't pursue _any_ of them, and
>go with the recognize-and-restrict approach I suggest at the beginning
>of this message.
>
>RjS


From nobody Thu Oct 23 11:24:13 2014
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Subject: Re: [sipcore] Fwd: New Version Notification for draft-sparks-sipcore-refer-explicit-subscription-02.txt
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This update seems good. I have one comment on section 4.3:

Using a *GRUU* doesn't make the URI hard to guess.
Using a *temporary GRUU* does.

	Thanks,
	Paul

On 10/21/14 3:52 PM, Robert Sparks wrote:
> fyi
>
>
> -------- Forwarded Message --------
> Subject: 	New Version Notification for
> draft-sparks-sipcore-refer-explicit-subscription-02.txt
> Date: 	Tue, 21 Oct 2014 12:26:36 -0700
> From: 	internet-drafts@ietf.org
> To: 	Robert Sparks <rjsparks@nostrum.com>, Robert Sparks
> <rjsparks@nostrum.com>
>
>
>
> A new version of I-D, draft-sparks-sipcore-refer-explicit-subscription-02.txt
> has been successfully submitted by Robert Sparks and posted to the
> IETF repository.
>
> Name:		draft-sparks-sipcore-refer-explicit-subscription
> Revision:	02
> Title:		Explicit Subscriptions for the REFER Method
> Document date:	2014-10-21
> Group:		Individual Submission
> Pages:		13
> URL:http://www.ietf.org/internet-drafts/draft-sparks-sipcore-refer-explicit-subscription-02.txt
> Status:https://datatracker.ietf.org/doc/draft-sparks-sipcore-refer-explicit-subscription/
> Htmlized:http://tools.ietf.org/html/draft-sparks-sipcore-refer-explicit-subscription-02
> Diff:http://www.ietf.org/rfcdiff?url2=draft-sparks-sipcore-refer-explicit-subscription-02
>
> Abstract:
>     The Session Initiation Protocol (SIP) REFER request, as defined by
>     RFC3515, triggers an implicit SIP-Specific Event Notification
>     framework subscription.  Conflating the start of the subscription
>     with handling the REFER request makes negotiating SUBSCRIBE
>     extensions impossible, and complicates avoiding SIP dialog sharing.
>     This document defines extensions to REFER to remove the implicit
>     subscription and, if desired, replace it with an explicit one.
>
>
>
>
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> The IETF Secretariat
>
>
>
>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>


From nobody Thu Oct 23 12:01:32 2014
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Subject: [sipcore] Agenda Items for SIPCORE at IETF91
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Now is the time to speak up if you want time on the SIPCORE agenda for 
the upcoming meeting. Please provide a rough outline of what you want to 
cover and how long you think you need to do so.

We have a one-hour session on Thursday, Nov 13.

	Thanks,
	Paul


From nobody Sun Oct 26 22:58:52 2014
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Subject: [sipcore] I-D Action: draft-ietf-sipcore-dns-dual-stack-01.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Session Initiation Protocol Core Working Group of the IETF.

        Title           : Locating Session Initiation Protocol (SIP) Servers in a Dual-Stack IP Network
        Authors         : Olle E. Johansson
                          Gonzalo Salgueiro
	Filename        : draft-ietf-sipcore-dns-dual-stack-01.txt
	Pages           : 7
	Date            : 2014-10-26

Abstract:
   RFC 3263 defines how a Session Initiation Protocol (SIP)
   implementation, given a SIP Uniform Resource Identifier (URI), should
   locate the next hop SIP server using Domain Name System (DNS)
   procedures.  As SIP networks increasingly transition from IPv4-only
   to dual-stack, a quality user experience must be ensured for dual-
   stack SIP implementations.  This document supplements the DNS
   procedures described in RFC 3263 for dual-stack SIP implementations
   and ensures that they properly align to the optimizations detailed by
   Happy Eyeballs.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-sipcore-dns-dual-stack/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-sipcore-dns-dual-stack-01

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-sipcore-dns-dual-stack-01


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


From nobody Mon Oct 27 03:34:25 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: Draft new: draft-holmberg-sipcore-auth-id-00
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Hi,

I've submitted a new draft, draft-holmberg-sipcore-auth-id-00.

The draft defines a new Authorization header field parameter, which can car=
ry a string value identifying an authorization server, based on a requireme=
nt from 3GPP (the mechanism is general, though).

The mechanism can be used together with the mechanism defined in draft-yuse=
f-sipcore-sip-oauth-01.

Regards,

Christer

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xmlns:m=3D"http://schemas.microsoft.com/office/2004/12/omml" xmlns=3D"http:=
//www.w3.org/TR/REC-html40">
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<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I&#8217;ve submitted a new draft, draft-holmberg-sip=
core-auth-id-00.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The draft defines a new Authorization header field p=
arameter, which can carry a string value identifying an authorization serve=
r, based on a requirement from 3GPP (the mechanism is general, though).<o:p=
></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The mechanism can be used together with the mechanis=
m defined in draft-yusef-sipcore-sip-oauth-01.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
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From nobody Mon Oct 27 07:02:58 2014
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Subject: Re: [sipcore] I-D Action: draft-sparks-sipcore-refer-clarifications-04.txt
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On 10/21/14 13:08, Paul Kyzivat wrote:
> I'd like some clarity on what "must use a gruu" means.
> Does this mean the contact must have a "gr" parameter, or only that it 
> must be globally routable? (Note that 3261 has always required contact 
> URIs to be globally routable.) The key difference is that with a "gr" 
> parameter the other end can verify that the uri has been asserted to 
> be globally routable.

I'm using GRUU as a shorthand for "the mechanism described in RFC5627" 
because it's easier to type. I think applying the term "GRUU" to other 
kinds of URIs, regardless of their routing properties, is going to muddy 
the waters, and make it nearly impossible to talk about the RFC5627 
mechanism.

In any case, the global routability of Contacts hasn't been true in a 
formal sense since the introduction of RFC3327, and hasn't been true in 
a practical sense since the introduction of NATs. We can't rely on it 
being true; and, in fact, it's going to be false in the vast majority of 
cases unless RFC5627 is employed.

/a


From nobody Mon Oct 27 07:17:45 2014
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-------- Forwarded Message --------
Subject: 	New Version Notification for 
draft-sparks-sipcore-refer-clarifications-05.txt
Date: 	Mon, 27 Oct 2014 07:13:11 -0700
From: 	internet-drafts@ietf.org
To: 	Adam Roach <adam@nostrum.com>, Adam Roach <adam@nostrum.com>, 
Robert Sparks <rjsparks@nostrum.com>, Robert Sparks <rjsparks@nostrum.com>



A new version of I-D, draft-sparks-sipcore-refer-clarifications-05.txt
has been successfully submitted by Robert Sparks and posted to the
IETF repository.

Name:		draft-sparks-sipcore-refer-clarifications
Revision:	05
Title:		Clarifications for the use of REFER with RFC6665
Document date:	2014-10-27
Group:		Individual Submission
Pages:		5
URL:            http://www.ietf.org/internet-drafts/draft-sparks-sipcore-refer-clarifications-05.txt
Status:         https://datatracker.ietf.org/doc/draft-sparks-sipcore-refer-clarifications/
Htmlized:       http://tools.ietf.org/html/draft-sparks-sipcore-refer-clarifications-05
Diff:           http://www.ietf.org/rfcdiff?url2=draft-sparks-sipcore-refer-clarifications-05

Abstract:
    An accepted SIP REFER method creates an implicit subscription using
    the SIP-Specific Event Notification Framework.  That framework was
    revised by RFC6665.  This document highlights the implications of the
    requirement changes in RFC6665, and updates the definition of the
    REFER method, RFC3515, to clarify and disambiguate the impact of
    those changes.

                                                                                   


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              draft-sparks-sipcore-refer-clarifications-05.txt</td>
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            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
            <td>Adam Roach <a class="moz-txt-link-rfc2396E" href="mailto:adam@nostrum.com">&lt;adam@nostrum.com&gt;</a>, Adam Roach
              <a class="moz-txt-link-rfc2396E" href="mailto:adam@nostrum.com">&lt;adam@nostrum.com&gt;</a>, Robert Sparks
              <a class="moz-txt-link-rfc2396E" href="mailto:rjsparks@nostrum.com">&lt;rjsparks@nostrum.com&gt;</a>, Robert Sparks
              <a class="moz-txt-link-rfc2396E" href="mailto:rjsparks@nostrum.com">&lt;rjsparks@nostrum.com&gt;</a></td>
          </tr>
        </tbody>
      </table>
      <br>
      <br>
      <pre>A new version of I-D, draft-sparks-sipcore-refer-clarifications-05.txt
has been successfully submitted by Robert Sparks and posted to the
IETF repository.

Name:		draft-sparks-sipcore-refer-clarifications
Revision:	05
Title:		Clarifications for the use of REFER with RFC6665
Document date:	2014-10-27
Group:		Individual Submission
Pages:		5
URL:            <a class="moz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-sparks-sipcore-refer-clarifications-05.txt">http://www.ietf.org/internet-drafts/draft-sparks-sipcore-refer-clarifications-05.txt</a>
Status:         <a class="moz-txt-link-freetext" href="https://datatracker.ietf.org/doc/draft-sparks-sipcore-refer-clarifications/">https://datatracker.ietf.org/doc/draft-sparks-sipcore-refer-clarifications/</a>
Htmlized:       <a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-sparks-sipcore-refer-clarifications-05">http://tools.ietf.org/html/draft-sparks-sipcore-refer-clarifications-05</a>
Diff:           <a class="moz-txt-link-freetext" href="http://www.ietf.org/rfcdiff?url2=draft-sparks-sipcore-refer-clarifications-05">http://www.ietf.org/rfcdiff?url2=draft-sparks-sipcore-refer-clarifications-05</a>

Abstract:
   An accepted SIP REFER method creates an implicit subscription using
   the SIP-Specific Event Notification Framework.  That framework was
   revised by RFC6665.  This document highlights the implications of the
   requirement changes in RFC6665, and updates the definition of the
   REFER method, RFC3515, to clarify and disambiguate the impact of
   those changes.

                                                                                  


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

The IETF Secretariat

</pre>
      <br>
    </div>
    <br>
  </body>
</html>

--------------000202080101050504010604--


From nobody Mon Oct 27 07:46:29 2014
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Date: Mon, 27 Oct 2014 10:45:31 -0400
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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To: Adam Roach <adam@nostrum.com>, sipcore@ietf.org
References: <7ECFBBA7034CE9ietf.shinji@gmail.com> <54469111.7020103@nostrum.com> <5446A130.9010602@alum.mit.edu> <544E5055.8040901@nostrum.com>
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Subject: Re: [sipcore] I-D Action: draft-sparks-sipcore-refer-clarifications-04.txt
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On 10/27/14 10:01 AM, Adam Roach wrote:
> On 10/21/14 13:08, Paul Kyzivat wrote:
>> I'd like some clarity on what "must use a gruu" means.
>> Does this mean the contact must have a "gr" parameter, or only that it
>> must be globally routable? (Note that 3261 has always required contact
>> URIs to be globally routable.) The key difference is that with a "gr"
>> parameter the other end can verify that the uri has been asserted to
>> be globally routable.
>
> I'm using GRUU as a shorthand for "the mechanism described in RFC5627"
> because it's easier to type. I think applying the term "GRUU" to other
> kinds of URIs, regardless of their routing properties, is going to muddy
> the waters, and make it nearly impossible to talk about the RFC5627
> mechanism.
>
> In any case, the global routability of Contacts hasn't been true in a
> formal sense since the introduction of RFC3327, and hasn't been true in
> a practical sense since the introduction of NATs. We can't rely on it
> being true; and, in fact, it's going to be false in the vast majority of
> cases unless RFC5627 is employed.

My concern is not with generic user endpoints, but with servers managed 
by enterprises or service providers. For instance a conference focus. 
These may have well known URIs, but they don't get them via registration.

RFC5627 does cover these - they are called self-made GRUUs. It does call 
for these to be marked with a "gr" parameter.

So then in practice, the requirement is that the contact have a "gr" 
parameter, and conform to RFC5627 for use of that.

	Thanks,
	Paul


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Subject: Re: [sipcore] Draft new: draft-holmberg-sipcore-auth-id-00
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> From: Christer Holmberg <christer.holmberg@ericsson.com>
> 
> I've submitted a new draft, draft-holmberg-sipcore-auth-id-00.

It would help if the Abstract was rewritten so that one did not need
to know the meanings of "eP-CSCF", "WAF", and "S-CSCF" to understand
what the mechanism can do.

> The draft defines a new Authorization header field parameter, which
> can carry a string value identifying an authorization server, based
> on a requirement from 3GPP (the mechanism is general, though).

The text says that the "authorization-entity" parameter carries "a
string value which represents the identity of an authorization
server".  You might want to expand on that.  I'm guessing it means
that any entity that wants to verify the authorization of the REGISTER
request should present the request contents to the specified
authorization server.

Dale


From nobody Tue Oct 28 01:29:28 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Dale R. Worley" <worley@ariadne.com>
Thread-Topic: [sipcore] Draft new: draft-holmberg-sipcore-auth-id-00
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Subject: Re: [sipcore] Draft new: draft-holmberg-sipcore-auth-id-00
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Hi,

>> I've submitted a new draft, draft-holmberg-sipcore-auth-id-00.
>
> It would help if the Abstract was rewritten so that one did not need to k=
now the meanings of "eP-CSCF", "WAF", and "S-CSCF" to understand what the m=
echanism can do.

I had very limited time to write the draft, so that's the reason it ended u=
p in this way. But, I DO agree with you, and that will be fixed in the next=
 version :)

>> The draft defines a new Authorization header field parameter, which=20
>> can carry a string value identifying an authorization server, based on=20
>> a requirement from 3GPP (the mechanism is general, though).
>
> The text says that the "authorization-entity" parameter carries "a string=
 value which represents the identity of an authorization server".  You migh=
t want to expand
> on that.  I'm guessing it means that any entity that wants to verify the =
authorization of the REGISTER request should present the request contents t=
o the specified=20
> authorization server.

I am not sure I understand what you mean by "present the request contents t=
o the specified authorization server". The identity is obtained from the au=
thorization server (or, the eP-CSCF knows it by configuration). The identit=
y is then provided to the S-CSCF (SIP registrar), which can use the informa=
tion for policy decisions.

Regards,

Christer


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Subject: Re: [sipcore] Draft new: draft-holmberg-sipcore-auth-id-00
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> From: Christer Holmberg <christer.holmberg@ericsson.com>

> >> The draft defines a new Authorization header field parameter, which 
> >> can carry a string value identifying an authorization server, based on 
> >> a requirement from 3GPP (the mechanism is general, though).
> >
> > The text says that the "authorization-entity" parameter carries "a
> > string value which represents the identity of an authorization
> > server".  You might want to expand on that.  I'm guessing it means
> > that any entity that wants to verify the authorization of the
> > REGISTER request should present the request contents to the
> > specified authorization server.
> 
> I am not sure I understand what you mean by "present the request
> contents to the specified authorization server". The identity is
> obtained from the authorization server (or, the eP-CSCF knows it by
> configuration). The identity is then provided to the S-CSCF (SIP
> registrar), which can use the information for policy decisions.

The general idea is this:  Suppose the reader has no idea how IMS does
things.  Within the SIP authorization model(s) presented in the RFCs,
what is the function of "authorization-entity"?  I assume that you
intend to extend the set of SIP authorization models; it's necessary
to describe, at least in outline, what the new authorization model is
and how "authorization-entity" is used in it so people can tell
whether they want to use it in their SIP implementations and how they
would do so.

As this draft is currently written, it seems to be "IMS needs this, so
we'll write that down and call it an RFC".  It needs considerable
clarification.

Dale


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Dale R. Worley" <worley@ariadne.com>
Thread-Topic: [sipcore] Draft new: draft-holmberg-sipcore-auth-id-00
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Subject: Re: [sipcore] Draft new: draft-holmberg-sipcore-auth-id-00
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Hi,

>>>> The draft defines a new Authorization header field parameter, which=20
>>>> can carry a string value identifying an authorization server, based=20
>>>> on a requirement from 3GPP (the mechanism is general, though).
>>>
>>> The text says that the "authorization-entity" parameter carries "a=20
>>> string value which represents the identity of an authorization=20
>>> server".  You might want to expand on that.  I'm guessing it means=20
>>> that any entity that wants to verify the authorization of the=20
>>> REGISTER request should present the request contents to the=20
>>> specified authorization server.
>>=20
>> I am not sure I understand what you mean by "present the request=20
>> contents to the specified authorization server". The identity is=20
>> obtained from the authorization server (or, the eP-CSCF knows it by=20
>> configuration). The identity is then provided to the S-CSCF (SIP=20
>> registrar), which can use the information for policy decisions.
>
> The general idea is this:  Suppose the reader has no idea how IMS does th=
ings.  Within the SIP authorization=20
> model(s) presented in the RFCs, what is the function of "authorization-en=
tity"?=20

The function is to provide the authorization server identity to the S-CSCF =
(registrar). The S-CSCF MAY use that information when authorizing a user. F=
or example, it can check whether the authorization server is authorized to =
provide IMS credentials to the user.

3GPP does not mandate the usage of the information, and does not specify th=
e policy procedures how it's used. 3GPP only requires the information to be=
 conveyed to the S-CSCF.

> I assume that you intend to extend the set of SIP authorization models; i=
t's necessary to describe, at least in outline, what the new authorization =
model is and how=20
> "authorization-entity" is used in it so people can tell whether they want=
 to use it in their SIP implementations and how they would do so.

The way the IMS core authorizes users as such is not changed.=20

The information is there so that IMS can use the information to check wheth=
er the authorization server is used, but that authorization server may serv=
e multiple users.=20

> As this draft is currently written, it seems to be "IMS needs this, so we=
'll write that down and call it an RFC".  It needs considerable clarificati=
on.

As always, we need to decide whether there is any interest for this outside=
 3GPP. But, I am happy to provide more information/clarification.

Regards,

Christer


From nobody Tue Oct 28 14:12:25 2014
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Subject: Re: [sipcore] Agenda Items for SIPCORE at IETF91
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I have now heard from three people with requests for agenda time.
Based on those requests, I've posted a *preliminary* agenda, at:

http://www.ietf.org/proceedings/91/agenda/agenda-91-sipcore

We only have a one hour slot, and there are requests for more time that 
that. For now I have assigned Robert his minimum requested time for the 
two REFER drafts. I consider them the highest priority, and if need be 
we will let them run longer.

For now I simply divided the remaining time among the other requests. 
But I don't expect it to stay that one. One of those is back for the 2nd 
time, with updates responding to comments in Toronto. It most likely 
will have priority over the others.

The other two are new. Rifaat's key derivation draft has had some list 
discussion. Christer's is newer, and has had just a little bit of list 
discussion.

It is probably counterproductive to give each draft too little time. So 
I'll be watching which ones have the most list discussion between now 
and the meeting, and will juggle the times accordingly.

	Thanks,
	Paul

On 10/23/14 3:01 PM, Paul Kyzivat wrote:
> Now is the time to speak up if you want time on the SIPCORE agenda for
> the upcoming meeting. Please provide a rough outline of what you want to
> cover and how long you think you need to do so.
>
> We have a one-hour session on Thursday, Nov 13.
>
>      Thanks,
>      Paul
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: [sipcore] Agenda Items for SIPCORE at IETF91
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Subject: Re: [sipcore] Agenda Items for SIPCORE at IETF91
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Hi,

Some comments on the pre-agenda:

First, as I think 10 (or shorter) minute slots at f2f meetings are not very=
 useful, I am willing to give up my slot (Authorization server identity), a=
nd let you allocate that time to something else. However, I'd like to have =
some off-line/list discussions on whether people think the solution could b=
e of general interest, or whether it's only of interest to 3GPP.

Second, while I do agree that Robert's draft has highest priority, do we re=
ally need f2f time for Explicit Subscriptions for REFER? Are there any open=
 issues (we discussed the forking issue, but based on the input from people=
 I believe we came to a conclusion on how to move forward)? Can't we just c=
all for adoption of the draft on the list?

Third, 5 minutes for Clarifications on use of REFER is just enough for Hadr=
iel to give his break-the-ice-joke-that-nobody-but-himself-understands :) A=
nd, as there has actually been quite a bit of discussions on that draft, I'=
d like the chairs to allocate more time for it. But, also in this case I th=
ink we should ask for adoption on the list.
=20
Thanks!

Christer

-----Original Message-----
From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Paul Kyzivat
Sent: 28. lokakuuta 2014 23:12
To: sipcore@ietf.org
Subject: Re: [sipcore] Agenda Items for SIPCORE at IETF91

I have now heard from three people with requests for agenda time.
Based on those requests, I've posted a *preliminary* agenda, at:

http://www.ietf.org/proceedings/91/agenda/agenda-91-sipcore

We only have a one hour slot, and there are requests for more time that tha=
t. For now I have assigned Robert his minimum requested time for the two RE=
FER drafts. I consider them the highest priority, and if need be we will le=
t them run longer.

For now I simply divided the remaining time among the other requests.=20
But I don't expect it to stay that one. One of those is back for the 2nd ti=
me, with updates responding to comments in Toronto. It most likely will hav=
e priority over the others.

The other two are new. Rifaat's key derivation draft has had some list disc=
ussion. Christer's is newer, and has had just a little bit of list discussi=
on.

It is probably counterproductive to give each draft too little time. So I'l=
l be watching which ones have the most list discussion between now and the =
meeting, and will juggle the times accordingly.

	Thanks,
	Paul

On 10/23/14 3:01 PM, Paul Kyzivat wrote:
> Now is the time to speak up if you want time on the SIPCORE agenda for=20
> the upcoming meeting. Please provide a rough outline of what you want=20
> to cover and how long you think you need to do so.
>
> We have a one-hour session on Thursday, Nov 13.
>
>      Thanks,
>      Paul
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

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


From nobody Wed Oct 29 12:04:26 2014
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Subject: Re: [sipcore] Agenda Items for SIPCORE at IETF91
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On 10/29/14 3:11 AM, Christer Holmberg wrote:
> Hi,
>
> Some comments on the pre-agenda:
>
> First, as I think 10 (or shorter) minute slots at f2f meetings are not very useful,

I agree. I was intending to fix that after some discussion.

> I am willing to give up my slot (Authorization server identity), and let you allocate that time to something else.

Thank you!

> However, I'd like to have some off-line/list discussions on whether people think the solution could be of general interest, or whether it's only of interest to 3GPP.

I haven't had time to review it yet. And now, if its not going to be on 
the agenda, I may not get to it until after the meeting.

You may have better luck getting attention *after* the meeting. 
(Realistically that probably means early next year.)

> Second, while I do agree that Robert's draft has highest priority, do we really need f2f time for Explicit Subscriptions for REFER? Are there any open issues (we discussed the forking issue, but based on the input from people I believe we came to a conclusion on how to move forward)? Can't we just call for adoption of the draft on the list?
> Third, 5 minutes for Clarifications on use of REFER is just enough for Hadriel to give his break-the-ice-joke-that-nobody-but-himself-understands :) And, as there has actually been quite a bit of discussions on that draft, I'd like the chairs to allocate more time for it. But, also in this case I think we should ask for adoption on the list.

Robert asked for at least 20 minutes, preferably 30, for both. The split 
between them was a swag by me.

	Thanks,
	Paul

> Thanks!
>
> Christer
>
> -----Original Message-----
> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: 28. lokakuuta 2014 23:12
> To: sipcore@ietf.org
> Subject: Re: [sipcore] Agenda Items for SIPCORE at IETF91
>
> I have now heard from three people with requests for agenda time.
> Based on those requests, I've posted a *preliminary* agenda, at:
>
> http://www.ietf.org/proceedings/91/agenda/agenda-91-sipcore
>
> We only have a one hour slot, and there are requests for more time that that. For now I have assigned Robert his minimum requested time for the two REFER drafts. I consider them the highest priority, and if need be we will let them run longer.
>
> For now I simply divided the remaining time among the other requests.
> But I don't expect it to stay that one. One of those is back for the 2nd time, with updates responding to comments in Toronto. It most likely will have priority over the others.
>
> The other two are new. Rifaat's key derivation draft has had some list discussion. Christer's is newer, and has had just a little bit of list discussion.
>
> It is probably counterproductive to give each draft too little time. So I'll be watching which ones have the most list discussion between now and the meeting, and will juggle the times accordingly.
>
> 	Thanks,
> 	Paul
>
> On 10/23/14 3:01 PM, Paul Kyzivat wrote:
>> Now is the time to speak up if you want time on the SIPCORE agenda for
>> the upcoming meeting. Please provide a rough outline of what you want
>> to cover and how long you think you need to do so.
>>
>> We have a one-hour session on Thursday, Nov 13.
>>
>>       Thanks,
>>       Paul
>>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>


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From: worley@ariadne.com (Dale R. Worley)
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Subject: Re: [sipcore] Draft new: draft-holmberg-sipcore-auth-id-00
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> From: Christer Holmberg <christer.holmberg@ericsson.com>

> The function is to provide the authorization server identity to the
> S-CSCF (registrar). The S-CSCF MAY use that information when
> authorizing a user. For example, it can check whether the
> authorization server is authorized to provide IMS credentials to the
> user.

When you say "provide the authorization server identity", so you mean
"provide the identity of the authorization server"?  And does that
mean that registrar/S-CSCF will use that as an input to its
authorization check (as basically a string, much like the username and
realm) or is it the DNS name of an authorization server that the
registrar is supposed to query to check the validity of the
credentials?

I think that you mean the former rather than the latter.

In that case, in what way does authorization-entity differ from the
username and realm?

> > I assume that you intend to extend the set of SIP authorization
> > models; it's necessary to describe, at least in outline, what the
> > new authorization model is and how "authorization-entity" is used
> > in it so people can tell whether they want to use it in their SIP
> > implementations and how they would do so.
> 
> The way the IMS core authorizes users as such is not changed. 

OK, but is the way that IMS core authorizes users different from the
usual SIP digest model?  If so, documenting that authorization method
could be useful for broader SIP usage.

> As always, we need to decide whether there is any interest for this
> outside 3GPP. But, I am happy to provide more
> information/clarification.

Granted.  But to determine that, people outside of 3GPP need to know
what the mechanism does, and that's not very clear yet.

Dale


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I've been negligent in pushing this draft along. There has been no 
action since Toronto, except that the draft was just refreshed to keep 
it from expiring.

I don't currently have it on the agenda for the Honolulu meeting because 
there has been no action and no request from the authors.

I would like to get this one wrapped up.

	Thanks,
	Paul

IIUC the following action is still pending.

On 6/11/14 4:50 PM, Vijay K. Gurbani wrote:
> On 06/11/2014 01:34 PM, Paul Kyzivat wrote:
>> SIPCORE people,
> [...]
>> Also, Ollie & Vijay - there is a TODO in the Introduction for the two of
>> you to sync up on relationship to 6157. We need to clear that up.
>
> ACK.  Will do.
>
> Thanks,
>
> - vijay


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From: "Gonzalo Salgueiro (gsalguei)" <gsalguei@cisco.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [sipcore] Finishing draft-ietf-sipcore-dns-dual-stack
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Date: Thu, 30 Oct 2014 19:04:55 +0000
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Cc: SIPCORE WG <sipcore@ietf.org>
Subject: Re: [sipcore] Finishing draft-ietf-sipcore-dns-dual-stack
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Hi Paul -=20

I concur. The recent inactivity is our fault (day job, etc) but we intend t=
o accelerate this effort as we move into the new year.  We recently reached=
 out to Vijay to collaborate on the only remaining open issue in the draft.=
  Once that has been settled we will send an updated version that I hope we=
 can consider for WGLC.

Thanks,

Gonzalo



> On Oct 30, 2014, at 2:43 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>=20
> I've been negligent in pushing this draft along. There has been no action=
 since Toronto, except that the draft was just refreshed to keep it from ex=
piring.
>=20
> I don't currently have it on the agenda for the Honolulu meeting because =
there has been no action and no request from the authors.
>=20
> I would like to get this one wrapped up.
>=20
> 	Thanks,
> 	Paul
>=20
> IIUC the following action is still pending.
>=20
> On 6/11/14 4:50 PM, Vijay K. Gurbani wrote:
>> On 06/11/2014 01:34 PM, Paul Kyzivat wrote:
>>> SIPCORE people,
>> [...]
>>> Also, Ollie & Vijay - there is a TODO in the Introduction for the two o=
f
>>> you to sync up on relationship to 6157. We need to clear that up.
>>=20
>> ACK.  Will do.
>>=20
>> Thanks,
>>=20
>> - vijay
>=20
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