
From nobody Thu Jun  2 07:57:30 2016
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Subject: [sipcore] sipcore - New Meeting Session Request for IETF 96
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A new meeting session request has just been submitted by Adam Roach, a Chair of the sipcore working group.


---------------------------------------------------------
Working Group Name: Session Initiation Protocol Core
Area Name: Applications and Real-Time Area
Session Requester: Adam Roach

Number of Sessions: 1
Length of Session(s):  1 Hour
Number of Attendees: 25
Conflicts to Avoid: 
 First Priority: netvc xrblock webpush stir mmusic rtcweb avtext perc avtcore dispatch




Special Requests:
  Cannot conflict with interledger bof.
---------------------------------------------------------


From nobody Tue Jun  7 12:31:22 2016
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Subject: [sipcore] Meeting in Berlin
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[as chair]

We've requested a short slot (one hour) at the upcoming IETF 96 meeting 
in Berlin for SIPCORE. We hope to help move along the AuthNZ discussion, 
and might have some conversations about the next steps for Happy 
Eyeballs. I would strongly encourage on-list discussion of these topics 
over the next few weeks.

If you have any additional topics that you hope to discuss at the 
face-to-face meeting, please let me and Jean know as soon as possible.

/a


From nobody Tue Jun  7 12:34:10 2016
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Subject: Re: [sipcore] Meeting in Berlin
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> On 07 Jun 2016, at 21:31, Adam Roach <adam@nostrum.com> wrote:
>=20
> [as chair]
>=20
> We've requested a short slot (one hour) at the upcoming IETF 96 =
meeting in Berlin for SIPCORE. We hope to help move along the AuthNZ =
discussion, and might have some conversations about the next steps for =
Happy Eyeballs. I would strongly encourage on-list discussion of these =
topics over the next few weeks.
>=20
> If you have any additional topics that you hope to discuss at the =
face-to-face meeting, please let me and Jean know as soon as possible.

I would like to start a discussion on half-legged SIP Outbound. This has =
been discussed in relation to SipConnect 2.0 and a few
earlier mails to this group. I will try to summarize my thoughts and the =
feedback I=E2=80=99ve seen so far very soon to prepare for a list
discussion prior to the meeting.

/O=


From nobody Tue Jun  7 13:48:45 2016
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From: Robert Sparks <rjsparks@nostrum.com>
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Date: Tue, 7 Jun 2016 15:48:33 -0500
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Subject: [sipcore] draft-sparks-sipcore-name-addr-guidance-00
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I took a stab at writing the document discussed in this thread.

The result is at 
https://datatracker.ietf.org/doc/draft-sparks-sipcore-name-addr-guidance/

Let me know where it's horribly broken.

RjS

On 4/2/16 8:02 AM, Robert Sparks wrote:
>
>
> On 4/1/16 10:56 AM, Paul Kyzivat wrote:
>> On 3/31/16 6:08 PM, Robert Sparks wrote:
>>> Again, I don't think errata is the right way to skin this.
>>> <http://mailarchive.ietf.org/arch/msg/sipcore/OZN1Eey88Ksex8fia20fCZH_8qE> 
>>>
>>
>> I agree that when in doubt a draft is the better path.
>>
>> But in this case, I don't think a revision to 3261 would cover all 
>> the other documents out there that define header fields with this 
>> problem. It would still be necessary to update each of those 
>> documents to adopt the change.
>>
>> But we could do one document that updates 3261 *and* each of the 
>> other affected documents.
> That's what my message said.
>> Then, in the future, any new header field where this should apply 
>> could simply normatively reference this document.
>>
>>     Thanks,
>>     Paul
>>
>>> RjS
>>>
>>> On 3/31/16 4:48 PM, Dale R. Worley wrote:
>>>> Brett Tate <brett@broadsoft.com> writes:
>>>>> I agree; that was why I sent the following email in October 2013 
>>>>> while
>>>>> sipcore was working on similar errata for RFC 3325.  I used that
>>>>> email and
>>>>> aspects of the RFC 3325 errata to produce the recent errata for RFC
>>>>> 3515,
>>>>> RFC 5002, RFC 5318, RFC 5360, and RFC 5502.  I have not check any
>>>>> headers
>>>>> register with IANA since October 2013.
>>>>>
>>>>> http://www.ietf.org/mail-archive/web/sipcore/current/msg05731.html
>>>> Why don't we put in an erratum that edits RFC 3261 section 20.10.  The
>>>> old text is:
>>>>
>>>>     Even if the "display-name" is empty, the "name-addr" form MUST be
>>>>     used if the "addr-spec" contains a comma, semicolon, or question
>>>>     mark.  There may or may not be LWS between the display-name and 
>>>> the
>>>>     "<".
>>>>
>>>> The new text adds:
>>>>
>>>>     In any header field whose value syntax is
>>>>         (name-addr / addr-spec) *(SEMI generic-param)
>>>>     or
>>>>         (name-addr / addr-spec) *(SEMI generic-param)
>>>>             *(COMMA (name-addr / addr-spec) *(SEMI generic-param))
>>>>     the "addr-spec" alternative may not be used if its value would
>>>>     contain a comma, semicolon, or question mark.
>>>>
>>>> In a sense, that's what everybody *thinks* RFC 3261 requires.
>>>>
>>>> Dale
>>>>
>>>> _______________________________________________
>>>> sipcore mailing list
>>>> sipcore@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>
>>
>> _______________________________________________
>> 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 Jun  8 06:15:00 2016
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] draft-sparks-sipcore-name-addr-guidance-00
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On 6/7/16 4:48 PM, Robert Sparks wrote:
> I took a stab at writing the document discussed in this thread.
>
> The result is at
> https://datatracker.ietf.org/doc/draft-sparks-sipcore-name-addr-guidance/
>
> Let me know where it's horribly broken.

It seems to do the job.

Do you think that the updates in section 3 can get away with not saying 
*where* that text is to be added in each of the documents?

	Thanks,
	Paul

> RjS
>
> On 4/2/16 8:02 AM, Robert Sparks wrote:
>>
>>
>> On 4/1/16 10:56 AM, Paul Kyzivat wrote:
>>> On 3/31/16 6:08 PM, Robert Sparks wrote:
>>>> Again, I don't think errata is the right way to skin this.
>>>> <http://mailarchive.ietf.org/arch/msg/sipcore/OZN1Eey88Ksex8fia20fCZH_8qE>
>>>>
>>>
>>> I agree that when in doubt a draft is the better path.
>>>
>>> But in this case, I don't think a revision to 3261 would cover all
>>> the other documents out there that define header fields with this
>>> problem. It would still be necessary to update each of those
>>> documents to adopt the change.
>>>
>>> But we could do one document that updates 3261 *and* each of the
>>> other affected documents.
>> That's what my message said.
>>> Then, in the future, any new header field where this should apply
>>> could simply normatively reference this document.
>>>
>>>     Thanks,
>>>     Paul
>>>
>>>> RjS
>>>>
>>>> On 3/31/16 4:48 PM, Dale R. Worley wrote:
>>>>> Brett Tate <brett@broadsoft.com> writes:
>>>>>> I agree; that was why I sent the following email in October 2013
>>>>>> while
>>>>>> sipcore was working on similar errata for RFC 3325.  I used that
>>>>>> email and
>>>>>> aspects of the RFC 3325 errata to produce the recent errata for RFC
>>>>>> 3515,
>>>>>> RFC 5002, RFC 5318, RFC 5360, and RFC 5502.  I have not check any
>>>>>> headers
>>>>>> register with IANA since October 2013.
>>>>>>
>>>>>> http://www.ietf.org/mail-archive/web/sipcore/current/msg05731.html
>>>>> Why don't we put in an erratum that edits RFC 3261 section 20.10.  The
>>>>> old text is:
>>>>>
>>>>>     Even if the "display-name" is empty, the "name-addr" form MUST be
>>>>>     used if the "addr-spec" contains a comma, semicolon, or question
>>>>>     mark.  There may or may not be LWS between the display-name and
>>>>> the
>>>>>     "<".
>>>>>
>>>>> The new text adds:
>>>>>
>>>>>     In any header field whose value syntax is
>>>>>         (name-addr / addr-spec) *(SEMI generic-param)
>>>>>     or
>>>>>         (name-addr / addr-spec) *(SEMI generic-param)
>>>>>             *(COMMA (name-addr / addr-spec) *(SEMI generic-param))
>>>>>     the "addr-spec" alternative may not be used if its value would
>>>>>     contain a comma, semicolon, or question mark.
>>>>>
>>>>> In a sense, that's what everybody *thinks* RFC 3261 requires.
>>>>>
>>>>> Dale
>>>>>
>>>>> _______________________________________________
>>>>> sipcore mailing list
>>>>> sipcore@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>>
>>>> _______________________________________________
>>>> sipcore mailing list
>>>> sipcore@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>>
>>>
>>> _______________________________________________
>>> 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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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Robert Sparks <rjsparks@nostrum.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: [sipcore] draft-sparks-sipcore-name-addr-guidance-00
Thread-Index: AQHRwP3+tTcgY2t210CSZdISPavKCJ/fpmYA
Date: Wed, 8 Jun 2016 13:40:48 +0000
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References: <87pouaqpxw.fsf@hobgoblin.ariadne.com> <56FD9FF7.7020207@nostrum.com> <56FE7E1C.50304@alum.mit.edu> <56FFC2ED.5030509@nostrum.com> <e84dc0df-d77c-8521-d797-c6b0393f6efd@nostrum.com>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/sipcore/Xf-anyQ49OtdGCxLwUXTPK9Snps>
Subject: Re: [sipcore] draft-sparks-sipcore-name-addr-guidance-00
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Hi,

If we want to be really clear, we should say *UNESCAPED* comma, question
mark or semicolon.

Because, the characters could appear escaped in most parts of a SIP-URI,
but I don=B9t think we should mandate usage of name-adde in such cases.

Regards,

Christer



On 07/06/16 23:48, "sipcore on behalf of Robert Sparks"
<sipcore-bounces@ietf.org on behalf of rjsparks@nostrum.com> wrote:

>I took a stab at writing the document discussed in this thread.
>
>The result is at=20
>https://datatracker.ietf.org/doc/draft-sparks-sipcore-name-addr-guidance/
>
>Let me know where it's horribly broken.
>
>RjS
>
>On 4/2/16 8:02 AM, Robert Sparks wrote:
>>
>>
>> On 4/1/16 10:56 AM, Paul Kyzivat wrote:
>>> On 3/31/16 6:08 PM, Robert Sparks wrote:
>>>> Again, I don't think errata is the right way to skin this.
>>>>=20
>>>><http://mailarchive.ietf.org/arch/msg/sipcore/OZN1Eey88Ksex8fia20fCZH_8
>>>>qE>=20
>>>>
>>>
>>> I agree that when in doubt a draft is the better path.
>>>
>>> But in this case, I don't think a revision to 3261 would cover all
>>> the other documents out there that define header fields with this
>>> problem. It would still be necessary to update each of those
>>> documents to adopt the change.
>>>
>>> But we could do one document that updates 3261 *and* each of the
>>> other affected documents.
>> That's what my message said.
>>> Then, in the future, any new header field where this should apply
>>> could simply normatively reference this document.
>>>
>>>     Thanks,
>>>     Paul
>>>
>>>> RjS
>>>>
>>>> On 3/31/16 4:48 PM, Dale R. Worley wrote:
>>>>> Brett Tate <brett@broadsoft.com> writes:
>>>>>> I agree; that was why I sent the following email in October 2013
>>>>>> while
>>>>>> sipcore was working on similar errata for RFC 3325.  I used that
>>>>>> email and
>>>>>> aspects of the RFC 3325 errata to produce the recent errata for RFC
>>>>>> 3515,
>>>>>> RFC 5002, RFC 5318, RFC 5360, and RFC 5502.  I have not check any
>>>>>> headers
>>>>>> register with IANA since October 2013.
>>>>>>
>>>>>> http://www.ietf.org/mail-archive/web/sipcore/current/msg05731.html
>>>>> Why don't we put in an erratum that edits RFC 3261 section 20.10.
>>>>>The
>>>>> old text is:
>>>>>
>>>>>     Even if the "display-name" is empty, the "name-addr" form MUST be
>>>>>     used if the "addr-spec" contains a comma, semicolon, or question
>>>>>     mark.  There may or may not be LWS between the display-name and
>>>>> the
>>>>>     "<".
>>>>>
>>>>> The new text adds:
>>>>>
>>>>>     In any header field whose value syntax is
>>>>>         (name-addr / addr-spec) *(SEMI generic-param)
>>>>>     or
>>>>>         (name-addr / addr-spec) *(SEMI generic-param)
>>>>>             *(COMMA (name-addr / addr-spec) *(SEMI generic-param))
>>>>>     the "addr-spec" alternative may not be used if its value would
>>>>>     contain a comma, semicolon, or question mark.
>>>>>
>>>>> In a sense, that's what everybody *thinks* RFC 3261 requires.
>>>>>
>>>>> Dale
>>>>>
>>>>> _______________________________________________
>>>>> sipcore mailing list
>>>>> sipcore@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>>
>>>> _______________________________________________
>>>> sipcore mailing list
>>>> sipcore@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>>
>>>
>>> _______________________________________________
>>> 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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To: sipcore@ietf.org
References: <87pouaqpxw.fsf@hobgoblin.ariadne.com> <56FD9FF7.7020207@nostrum.com> <56FE7E1C.50304@alum.mit.edu> <56FFC2ED.5030509@nostrum.com> <e84dc0df-d77c-8521-d797-c6b0393f6efd@nostrum.com> <9909848d-5035-2d46-f9e4-2e8e72a343b6@alum.mit.edu>
From: Robert Sparks <rjsparks@nostrum.com>
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Subject: Re: [sipcore] draft-sparks-sipcore-name-addr-guidance-00
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On 6/8/16 8:14 AM, Paul Kyzivat wrote:
> On 6/7/16 4:48 PM, Robert Sparks wrote:
>> I took a stab at writing the document discussed in this thread.
>>
>> The result is at
>> https://datatracker.ietf.org/doc/draft-sparks-sipcore-name-addr-guidance/ 
>>
>>
>> Let me know where it's horribly broken.
>
> It seems to do the job.
>
> Do you think that the updates in section 3 can get away with not 
> saying *where* that text is to be added in each of the documents?
Yes.

I see nothing but trouble in trying to place them more specifically 
given what they do, and I don't see how they can be misinterpreted 
without such placement.

>
>     Thanks,
>     Paul
>
>> RjS
>>
>> On 4/2/16 8:02 AM, Robert Sparks wrote:
>>>
>>>
>>> On 4/1/16 10:56 AM, Paul Kyzivat wrote:
>>>> On 3/31/16 6:08 PM, Robert Sparks wrote:
>>>>> Again, I don't think errata is the right way to skin this.
>>>>> <http://mailarchive.ietf.org/arch/msg/sipcore/OZN1Eey88Ksex8fia20fCZH_8qE> 
>>>>>
>>>>>
>>>>
>>>> I agree that when in doubt a draft is the better path.
>>>>
>>>> But in this case, I don't think a revision to 3261 would cover all
>>>> the other documents out there that define header fields with this
>>>> problem. It would still be necessary to update each of those
>>>> documents to adopt the change.
>>>>
>>>> But we could do one document that updates 3261 *and* each of the
>>>> other affected documents.
>>> That's what my message said.
>>>> Then, in the future, any new header field where this should apply
>>>> could simply normatively reference this document.
>>>>
>>>>     Thanks,
>>>>     Paul
>>>>
>>>>> RjS
>>>>>
>>>>> On 3/31/16 4:48 PM, Dale R. Worley wrote:
>>>>>> Brett Tate <brett@broadsoft.com> writes:
>>>>>>> I agree; that was why I sent the following email in October 2013
>>>>>>> while
>>>>>>> sipcore was working on similar errata for RFC 3325.  I used that
>>>>>>> email and
>>>>>>> aspects of the RFC 3325 errata to produce the recent errata for RFC
>>>>>>> 3515,
>>>>>>> RFC 5002, RFC 5318, RFC 5360, and RFC 5502.  I have not check any
>>>>>>> headers
>>>>>>> register with IANA since October 2013.
>>>>>>>
>>>>>>> http://www.ietf.org/mail-archive/web/sipcore/current/msg05731.html
>>>>>> Why don't we put in an erratum that edits RFC 3261 section 
>>>>>> 20.10.  The
>>>>>> old text is:
>>>>>>
>>>>>>     Even if the "display-name" is empty, the "name-addr" form 
>>>>>> MUST be
>>>>>>     used if the "addr-spec" contains a comma, semicolon, or question
>>>>>>     mark.  There may or may not be LWS between the display-name and
>>>>>> the
>>>>>>     "<".
>>>>>>
>>>>>> The new text adds:
>>>>>>
>>>>>>     In any header field whose value syntax is
>>>>>>         (name-addr / addr-spec) *(SEMI generic-param)
>>>>>>     or
>>>>>>         (name-addr / addr-spec) *(SEMI generic-param)
>>>>>>             *(COMMA (name-addr / addr-spec) *(SEMI generic-param))
>>>>>>     the "addr-spec" alternative may not be used if its value would
>>>>>>     contain a comma, semicolon, or question mark.
>>>>>>
>>>>>> In a sense, that's what everybody *thinks* RFC 3261 requires.
>>>>>>
>>>>>> Dale
>>>>>>
>>>>>> _______________________________________________
>>>>>> sipcore mailing list
>>>>>> sipcore@ietf.org
>>>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>>>
>>>>> _______________________________________________
>>>>> sipcore mailing list
>>>>> sipcore@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Date: Thu, 9 Jun 2016 13:59:15 -0400
Message-ID: <CAGL6epKiY_Xi_D+J8QnTfoEyaDU1=jfcbhVFAMY_v3tkQjAhug@mail.gmail.com>
To: "Peterson, Jon" <jon.peterson@neustar.biz>
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Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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--001a114413c04003c80534dc2e30
Content-Type: text/plain; charset=UTF-8

Hi Jon,

Sorry for the delayed reply.

Let's take the video conference example, and discuss the possible ways for the
conference server to provide service.


         Trust Domain 1              :    Internet     :  Trust Domain 2
                                     :                 :
                      +--------+     :                 :
                      | IdP    |     :                 :
        +-------------| AuthZ  |     :                 :
        |             |        |     :                 :
        |             +--------+     :                 :
        |                 |          :                 :
        |                 |          :                 :
    +--------+        +--------+     :                 :     +--------+
    |        |        | SIP    |     :                 :     | SIP App|
    |   UA   |--------| Proxy  |-----:-----------------:-----| Video  |
    |        |        |        |     :                 :     | Conference
    +--------+        +--------+     :                 :     +--------+
                                     :                 :

Possible solutions:

1. Conference Server acts as an AuthN and AuthZ server. Requests to
the server must provide an identity and credentials, and services will
be provided based on locally stored information associated with
the identity provided in the request.

2. Conference Server acts as an AuthZ server, but delegates the AuthN
part. Requests to the server must provide an identity token, and services
will be provided based on locally stored information associated with
the identity provided in the token.

3. Conference Server delegates both AuthN and AuthZ parts. Requests to
the server must provide an access token, which might or might not
include information about the user requesting the service, and services
will be provided based on information provided in the access token or as
a result of an introspection step using the access token.


Now, let's look at the 3rd option a bit more, and assume that the
server provides the following types of conference services:
1. audio conference
2. audio/video conference
3. web conference

or some combination of the above.

Let's also assume that the server allows control over the number
of participants allowed in each one of these conference types or
combinations.


Here is one way that I envisioned we could use to address this use case:

When an enterprise user in the 1st trust domain wants to register with
the SIP Proxy, it will have to authenticate to the IdP and as a result
either the UA or the Proxy (depends on the UA's capabilities) will be
provided with an AuthN and AuthZ tokens. This process could involve an
interim step to allow only the proxy to get access to the tokens, or the
tokens could be sent to the UA which will then use them to communicate with
the proxy.

The AuthN token is the identity of the user, and the AuthZ token has
all the services granted to the user by the IdP/AuthZ Service.

When the enterprise user needs access to the conference service, the
UA will send an INVITE to the Proxy, and will include the AuthZ token if it
is in possession of the tokens; otherwise, the proxy will add the
appropriate token to the outgoing INVITE sent to the conference server.

The scope of the AuthZ token might grant the user access to
multiple services; to avoid sending that AuthZ token to all servers
providing services granted to the user, the proxy could obtain a different
token with limited scope that is specific to the service being provided
(conference services in this case); there is a standard Token Exchange
mechanism for that.

The AuthZ token will include information about the type of service
allowed (audio, audio/video, etc), and the limit on the number of
participants. The AuthZ token might or might not have any information about
the user requesting the service, which depends on the level of privacy
needed.

When the conference server receives this token in the SIP INVITE, all
it has to do is validate that the token was signed by an IdP that it has
a trust relationship with, and provide the service requested by the
AuthZ token. The conference server does not need the AuthN token to be able
to provide a service.

Regards,
 Rifaat


On Wed, May 25, 2016 at 3:06 PM, Peterson, Jon <jon.peterson@neustar.biz>
wrote:

>
> Also, if we decided to separate the AuthN and AuthZ, then we might need to
> carry more than one token.
>
>
> I could see an argument that there is a need to carry the identity of the
> user in one header, and a reference to a place where you can get attributes
> about the originating user in another header, say. That's more or less what
> we envisioned the last time we went down this path, which led to the
> RFC4484 requirements, largely informed by SAML (which I seem to recall 3GPP
> was interested in at the time).
>
> But it's no accident that the work on RFC4484 never went beyond
> requirements. Today, I think that much of the useful work that could be
> done on the authorization question is in identifying which SIP
> authorization policies are likely to depend on attributes as opposed to
> just the identity of the end user. While we can readily conceive of
> applications adjunct to communications that would require attributes, the
> challenge has been justifying why you would invoke those applications with
> SIP rather than, say, HTTP.  The fundamental semantics of a SIP INVITE have
> remained pretty close to those of placing a telephone call.  The
> authorization policies of a telephone call are really different than that
> of a web service, say, both in terms of how and why they are invoked, what
> kinds of parties are involved, what forms redirection or cross-domain
> interaction can occur. With appropriate caveats for other methods (notably
> MESSAGE, SUBSCRIBE/NOTIFY. INFO), SIP requests have authorization policy
> requirements that look a lot like call treatment policies.
>
> That much said, I am kind of interested in call treatment policies, and
> their interaction with identity information. But just saying that there
> might be a video conference service in the cloud, say, doesn't shed a lot
> of light for me. If there's a video conference, I'd ordinarily think you
> just call into it. If the user who opens the conference bridge needs to be
> charged, that's identity information - which in its usual form will convey
> the individual user and their domain, like user@domain. If you need more
> complex semantics than that, it's probably because you're doing something
> more complicated than just calling in to the conference. But it's less
> clear to me why you'd use SIP for such cases. Drilling down deeper into
> these use cases would help me understand where we're going, anyway - the
> use cases in RFC4484 Section 4, which largely involve billing (including
> when dealing with various protocol gateways) and resource management (both
> preemption and priority), ultimately didn't seem sufficient to motivate
> protocol work.
>
> Another issues in the need, in some cases, to avoid sending the token to
> the UA; in this case there is a need for an interim step to get the
> token(s) to the SIP proxy.
>
>
> If this information is carried in headers, then presumably it could be
> added, consumed, subtracted, or whatever by any relevant intermediaries.
>
> Jon Peterson
> Neustar, Inc.
>

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<div dir=3D"ltr"><font face=3D"monospace, monospace">Hi Jon,</font><div><fo=
nt face=3D"monospace, monospace"><br></font></div><div><font face=3D"monosp=
ace, monospace">Sorry for the delayed reply.</font></div><div><font face=3D=
"monospace, monospace"><br></font></div><div><div><div class=3D"gmail_extra=
"><div class=3D"gmail_quote"><div><font face=3D"monospace, monospace">Let&#=
39;s take the video conference example, and discuss the possible ways for=
=C2=A0</font><span style=3D"font-family:monospace,monospace">the conference=
 server to provide service.</span></div><div><font face=3D"monospace, monos=
pace"><br></font></div><div><font face=3D"monospace, monospace"><br></font>=
</div><div><font face=3D"monospace, monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0Trust Domain 1 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0: =C2=
=A0 =C2=A0Internet =C2=A0 =C2=A0 : =C2=A0Trust Domain 2</font></div><div><f=
ont face=3D"monospace, monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0=
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
:</font></div><div><font face=3D"monospace, monospace">=C2=A0 =C2=A0 =C2=A0=
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 +--------+ =C2=A0 =
=C2=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :</font></=
div><div><font face=3D"monospace, monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | IdP =C2=A0 =C2=A0| =C2=
=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :</fon=
t></div><div><font face=3D"monospace, monospace">=C2=A0 =C2=A0 =C2=A0 =C2=
=A0 +-------------| AuthZ =C2=A0| =C2=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :</font></div><div><font face=3D"monospace,=
 monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0 =C2=A0| =C2=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :</font></div><div><font face=3D"mon=
ospace, monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 +--------+ =C2=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 :</font></div><div><font face=3D"monospace, mon=
ospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0: =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :</font></div><div><font face=3D"=
monospace, monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0: =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :</font></div><div>=
<font face=3D"monospace, monospace">=C2=A0 =C2=A0 +--------+ =C2=A0 =C2=A0 =
=C2=A0 =C2=A0+--------+ =C2=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 : =C2=A0 =C2=A0 +--------+</font></div><div><font face=
=3D"monospace, monospace">=C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0 =C2=A0| =C2=
=A0 =C2=A0 =C2=A0 =C2=A0| SIP =C2=A0 =C2=A0| =C2=A0 =C2=A0 : =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : =C2=A0 =C2=A0 | SIP App|</font>=
</div><div><font face=3D"monospace, monospace">=C2=A0 =C2=A0 | =C2=A0 UA =
=C2=A0 |--------| Proxy =C2=A0|-----:-----------------:-----| Video =C2=A0|=
</font></div><div><font face=3D"monospace, monospace">=C2=A0 =C2=A0 | =C2=
=A0 =C2=A0 =C2=A0 =C2=A0| =C2=A0 =C2=A0 =C2=A0 =C2=A0| =C2=A0 =C2=A0 =C2=A0=
 =C2=A0| =C2=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 : =C2=A0 =C2=A0 | Conference</font></div><div><font face=3D"monospac=
e, monospace">=C2=A0 =C2=A0 +--------+ =C2=A0 =C2=A0 =C2=A0 =C2=A0+--------=
+ =C2=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :=
 =C2=A0 =C2=A0 +--------+</font></div><div><font face=3D"monospace, monospa=
ce">=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0: =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :</font></div><div><font face=
=3D"monospace, monospace"><br></font></div><div><font face=3D"monospace, mo=
nospace">Possible solutions:</font></div><div><font face=3D"monospace, mono=
space"><br></font></div><div><font face=3D"monospace, monospace">1. Confere=
nce Server acts as an AuthN and AuthZ server. Requests to the=C2=A0server m=
ust provide an identity and credentials, and services will be=C2=A0provided=
 based on locally stored information associated with the=C2=A0identity prov=
ided in the request.</font></div><div><font face=3D"monospace, monospace"><=
br></font></div><div><font face=3D"monospace, monospace">2. Conference Serv=
er acts as an AuthZ server, but delegates the AuthN part.=C2=A0Requests to =
the server must provide an identity token, and services will=C2=A0be provid=
ed based on locally stored information associated with the=C2=A0identity pr=
ovided in the token.</font></div><div><font face=3D"monospace, monospace"><=
br></font></div><div><font face=3D"monospace, monospace">3. Conference Serv=
er delegates both AuthN and AuthZ parts. Requests to the=C2=A0server must p=
rovide an access token, which might or might not include=C2=A0information a=
bout the user requesting the service, and services will=C2=A0be provided ba=
sed on information provided in the access token or as a=C2=A0result of an i=
ntrospection step using the access token.</font></div><div><font face=3D"mo=
nospace, monospace"><br></font></div><div><font face=3D"monospace, monospac=
e"><br></font></div><div><font face=3D"monospace, monospace">Now, let&#39;s=
 look at the 3rd option a bit more, and assume that the server=C2=A0provide=
s the following types of conference services:</font></div><div><font face=
=3D"monospace, monospace">1. audio conference</font></div><div><font face=
=3D"monospace, monospace">2. audio/video conference</font></div><div><font =
face=3D"monospace, monospace">3. web conference</font></div><div><font face=
=3D"monospace, monospace"><br></font></div><div><font face=3D"monospace, mo=
nospace">or some=C2=A0combination=C2=A0of the above.</font></div><div><font=
 face=3D"monospace, monospace"><br></font></div><div><font face=3D"monospac=
e, monospace">Let&#39;s also assume that the server allows control over the=
 number of=C2=A0participants allowed in each one of these conference types =
or combinations.</font></div><div><font face=3D"monospace, monospace"><br><=
/font></div><div><font face=3D"monospace, monospace"><br></font></div><div>=
<font face=3D"monospace, monospace">Here is one way that I envisioned we co=
uld use to address this use case:</font></div><div><font face=3D"monospace,=
 monospace"><br></font></div></div></div></div><font face=3D"monospace, mon=
ospace">When an enterprise user in the 1st trust domain wants to register w=
ith the=C2=A0SIP Proxy, it will have to authenticate to the IdP and as a re=
sult either=C2=A0the UA or the Proxy (depends on the UA&#39;s capabilities)=
 will be provided=C2=A0with an AuthN and AuthZ tokens. This process could i=
nvolve an interim step=C2=A0to allow only the proxy to get access to the to=
kens, or the tokens could be=C2=A0sent to the UA which will then use them t=
o communicate with the proxy.</font><div><font face=3D"monospace, monospace=
"><br>The AuthN token is the identity of the user, and the AuthZ token has =
all=C2=A0the services granted to the user by the IdP/AuthZ Service.<br><br>=
When the enterprise user needs access to the conference service, the UA=C2=
=A0will send an INVITE to the Proxy, and will include the AuthZ token if it=
 is=C2=A0in possession of the tokens; otherwise, the proxy will add the app=
ropriate=C2=A0token to the outgoing INVITE sent to the conference server.<b=
r><br>The scope of the AuthZ token might grant the user access to multiple=
=C2=A0services; to avoid sending that AuthZ token to all servers providing=
=C2=A0services granted to the user, the proxy could obtain a different toke=
n with=C2=A0limited scope that is specific to the service being provided (c=
onference=C2=A0services in this case); there is a standard Token Exchange m=
echanism for=C2=A0that.<br><br>The AuthZ token will include information abo=
ut the type of service allowed=C2=A0(audio, audio/video, etc), and the limi=
t on the number of participants.=C2=A0The AuthZ token might or might not ha=
ve any information about the user=C2=A0requesting the service, which depend=
s on the level of privacy needed.<br><br>When the conference server receive=
s this token in the SIP INVITE, all it=C2=A0has to do is validate that the =
token was signed by an IdP that it has a=C2=A0trust relationship with, and =
provide the service requested by the AuthZ=C2=A0token. The conference serve=
r does not need the AuthN token to be able to=C2=A0provide a service.</font=
><div><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div><font face=
=3D"monospace, monospace"><br></font></div><div><font face=3D"monospace, mo=
nospace">Regards,</font></div><div><font face=3D"monospace, monospace">=C2=
=A0Rifaat</font></div><div><font face=3D"monospace, monospace"><br></font><=
/div></div></div></div></div></div></div><div class=3D"gmail_extra"><br><di=
v class=3D"gmail_quote">On Wed, May 25, 2016 at 3:06 PM, Peterson, Jon <spa=
n dir=3D"ltr">&lt;<a href=3D"mailto:jon.peterson@neustar.biz" target=3D"_bl=
ank">jon.peterson@neustar.biz</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 style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif"><span class=3D"">
<span>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
</div>
</div>
</div>
</span><span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>Also, if we decided to separate the AuthN and AuthZ, then we might nee=
d to carry more than one token.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span><div>I could see an argument that there is a need to carry the ident=
ity of the user in one header, and a reference to a place where you can get=
 attributes about the originating user in another header, say. That&#39;s m=
ore or less what we envisioned the last time
 we went down this path, which led to the RFC4484 requirements, largely inf=
ormed by SAML (which I seem to recall 3GPP was interested in at the time).<=
/div>
<div><br>
</div>
<div>But it&#39;s no accident that the work on RFC4484 never went beyond re=
quirements. Today, I think that much of the useful work that could be done =
on the authorization question is in identifying which SIP authorization pol=
icies are likely to depend on attributes
 as opposed to just the identity of the end user. While we can readily conc=
eive of applications adjunct to communications that would require attribute=
s, the challenge has been justifying why you would invoke those application=
s with SIP rather than, say, HTTP.
 =C2=A0The fundamental semantics of a SIP INVITE have remained pretty close=
 to those of placing a telephone call.=C2=A0 The authorization policies of =
a telephone call are really different than that of a web service, say, both=
 in terms of how and why they are invoked,
 what kinds of parties are involved, what forms redirection or cross-domain=
 interaction can occur. With appropriate caveats for other methods (notably=
 MESSAGE, SUBSCRIBE/NOTIFY. INFO), SIP requests have authorization policy r=
equirements that look a lot like
 call treatment policies.</div>
<div><br>
</div>
<div>That much said, I am kind of interested in call treatment policies, an=
d their interaction with identity information. But just saying that there m=
ight be a video conference service in the cloud, say, doesn&#39;t shed a lo=
t of light for me. If there&#39;s a video
 conference, I&#39;d ordinarily think you just call into it. If the user wh=
o opens the conference bridge needs to be charged, that&#39;s identity info=
rmation - which in its usual form will convey the individual user and their=
 domain, like user@domain. If you need more
 complex semantics than that, it&#39;s probably because you&#39;re doing so=
mething more complicated than just calling in to the conference. But it&#39=
;s less clear to me why you&#39;d use SIP for such cases. Drilling down dee=
per into these use cases would help me understand
 where we&#39;re going, anyway - the use cases in RFC4484 Section 4, which =
largely involve billing (including when dealing with various protocol gatew=
ays) and resource management (both preemption and priority), ultimately did=
n&#39;t seem sufficient to motivate protocol
 work.</div><span class=3D"">
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>Another issues in the need, in some cases, to avoid sending the token =
to the UA; in this case there is a need for an interim step to get the toke=
n(s) to the SIP proxy.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span><div>If this information is carried in headers, then presumably it c=
ould be added, consumed, subtracted, or whatever by any relevant intermedia=
ries.=C2=A0</div><span class=3D"HOEnZb"><font color=3D"#888888">
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
</font></span></div>

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

--001a114413c04003c80534dc2e30--


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Subject: [sipcore] Half-outbound thoughts
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Friends,
I=E2=80=99ve tried to collect my thoughts about TLS and Outbound in a =
small brainstorm-presentation.

http://www.slideshare.net/oej/sip-half-outbound-random-notes

The chairs have alerted me to the fact that this may not be a SIPcore =
issue which I agree with, but I think this is the proper starting point =
for a general discussion between SIP people before we either drop the =
issue or drag it into dispatch. My apologies to the chairs for using =
this attractive forum :-)

In short:

- very few seems to implement full outbound
- some parts of outbound seems used
- which parts is not documented
- TLS between UA and server has a clear problem if outbound is not used

I am toying with an idea for a simple extension that puts the =
connection/flow management on the UA,
allows the server to reuse the inbound TLS connection (without a client =
cert) after auth for outbound
requests and also allows the server to remove associations in the =
location database at connection close.

I think this documents what=E2=80=99s in use and formalises it with an =
extension tag.

Please check my presentation and give me feedback.

Thanks!
/O


From nobody Fri Jun 10 02:52:22 2016
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Subject: [sipcore] SIPS must die, die, die
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Friends,

I think it=E2=80=99s time to kill the SIPS: uri. I know that some of you =
thought it was dead, but it is not the case.

It doesn=E2=80=99t add anything, personally I don=E2=80=99t see it in =
use anywhere and it confuses a lot of discussions about TLS usage in =
SIP.

We need to see what happens if we remove SIPS: from the SIP playground - =
what=E2=80=99s missing?

Questions arise:

- Can we live without SIPS: and without =E2=80=9C;transport=3DTLS=E2=80=9D=
 or do we need to take the transport parameter back?
- Can my TLS connection reuse idea (previous mail) solve UA to edge =
proxy issues?
- Do we need TLS transport in VIA headers?
- Are we going to have to prepare for DTLS transport too? (See Dale =
Worley=E2=80=99s previous emails) ?

I think it=E2=80=99s time to spend some time with TLS in SIPcore again. =
What do you think?

/O


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Olle E. Johansson" <oej@edvina.net>, SIPCORE <sipcore@ietf.org>
Thread-Topic: [sipcore] Half-outbound thoughts
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Subject: Re: [sipcore] Half-outbound thoughts
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Hi O,

A couple of questions on the slides:

Q1: Number of flows

You claim that Outbound mandates at least two flows.

I can find text saying that a UA must, as a minimum, support a set of two
outbound proxy URIs, and while the whole idea of Outbound is having
multiple flows I don=B9t find any text saying that one must configure two
(or more) outbound proxy URIs.


Q2: Flow-management

Slide #14 says =B3No flow-management=B2. What is meant by that? That the
keep-olives aren=B9t needed?


Regards,

Christer



On 10/06/16 12:46, "sipcore on behalf of Olle E. Johansson"
<sipcore-bounces@ietf.org on behalf of oej@edvina.net> wrote:

>Friends,
>I=B9ve tried to collect my thoughts about TLS and Outbound in a small
>brainstorm-presentation.
>
>http://www.slideshare.net/oej/sip-half-outbound-random-notes
>
>The chairs have alerted me to the fact that this may not be a SIPcore
>issue which I agree with, but I think this is the proper starting point
>for a general discussion between SIP people before we either drop the
>issue or drag it into dispatch. My apologies to the chairs for using this
>attractive forum :-)
>
>In short:
>
>- very few seems to implement full outbound
>- some parts of outbound seems used
>- which parts is not documented
>- TLS between UA and server has a clear problem if outbound is not used
>
>I am toying with an idea for a simple extension that puts the
>connection/flow management on the UA,
>allows the server to reuse the inbound TLS connection (without a client
>cert) after auth for outbound
>requests and also allows the server to remove associations in the
>location database at connection close.
>
>I think this documents what=B9s in use and formalises it with an extension
>tag.
>
>Please check my presentation and give me feedback.
>
>Thanks!
>/O
>
>_______________________________________________
>sipcore mailing list
>sipcore@ietf.org
>https://www.ietf.org/mailman/listinfo/sipcore


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Subject: Re: [sipcore] Half-outbound thoughts
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> On 10 Jun 2016, at 12:10, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>=20
> Hi O,
>=20
> A couple of questions on the slides:
>=20
> Q1: Number of flows
>=20
> You claim that Outbound mandates at least two flows.
>=20
> I can find text saying that a UA must, as a minimum, support a set of =
two
> outbound proxy URIs, and while the whole idea of Outbound is having
> multiple flows I don=C4=85t find any text saying that one must =
configure two
> (or more) outbound proxy URIs.
We=E2=80=99ve discussed this many times at SIPit and have parsed TWO as =
a=20
must. I think you may be right though, that two is not required to run,
but the client must support using it. My failure.

Additional quote:

"If the set has more than
   one URI, the UAC MUST send a REGISTER request to at least two of the
   default outbound proxies from the set. =E2=80=9C

I will update the slides
>=20
>=20
> Q2: Flow-management
>=20
> Slide #14 says =C5=82No flow-management=CB=9B. What is meant by that? =
That the
> keep-olives aren=C4=85t needed?
>=20
That=E2=80=99s very badly written by me. We need flow management, but =
not in
the outbound-style, with flow tokens and all that. Will update.

Thank you for very good feedback!

/O
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
> On 10/06/16 12:46, "sipcore on behalf of Olle E. Johansson"
> <sipcore-bounces@ietf.org on behalf of oej@edvina.net> wrote:
>=20
>> Friends,
>> I=C4=85ve tried to collect my thoughts about TLS and Outbound in a =
small
>> brainstorm-presentation.
>>=20
>> http://www.slideshare.net/oej/sip-half-outbound-random-notes
>>=20
>> The chairs have alerted me to the fact that this may not be a SIPcore
>> issue which I agree with, but I think this is the proper starting =
point
>> for a general discussion between SIP people before we either drop the
>> issue or drag it into dispatch. My apologies to the chairs for using =
this
>> attractive forum :-)
>>=20
>> In short:
>>=20
>> - very few seems to implement full outbound
>> - some parts of outbound seems used
>> - which parts is not documented
>> - TLS between UA and server has a clear problem if outbound is not =
used
>>=20
>> I am toying with an idea for a simple extension that puts the
>> connection/flow management on the UA,
>> allows the server to reuse the inbound TLS connection (without a =
client
>> cert) after auth for outbound
>> requests and also allows the server to remove associations in the
>> location database at connection close.
>>=20
>> I think this documents what=C4=85s in use and formalises it with an =
extension
>> tag.
>>=20
>> Please check my presentation and give me feedback.
>>=20
>> Thanks!
>> /O
>>=20
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From nobody Fri Jun 10 05:44:31 2016
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References: <F3615DC1-4866-4C5D-8E5D-D3E1BAE6984E@edvina.net> <D3806BAB.A8A6%christer.holmberg@ericsson.com> <53B14A35-EBF0-4FB1-A20A-59B5E0F7F63A@edvina.net>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: SIPCORE <sipcore@ietf.org>, Olle E Johansson <oej@edvina.net>
Subject: Re: [sipcore] Half-outbound thoughts
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> On 10 Jun 2016, at 13:04, Olle E. Johansson <oej@edvina.net> wrote:
>=20
>=20
>> On 10 Jun 2016, at 12:10, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>>=20
>> Hi O,
>>=20
>> A couple of questions on the slides:
>>=20
>> Q1: Number of flows
>>=20
>> You claim that Outbound mandates at least two flows.
>>=20
>> I can find text saying that a UA must, as a minimum, support a set of =
two
>> outbound proxy URIs, and while the whole idea of Outbound is having
>> multiple flows I don=C4=85t find any text saying that one must =
configure two
>> (or more) outbound proxy URIs.
> We=E2=80=99ve discussed this many times at SIPit and have parsed TWO =
as a=20
> must. I think you may be right though, that two is not required to =
run,
> but the client must support using it. My failure.
>=20
> Additional quote:
>=20
> "If the set has more than
>   one URI, the UAC MUST send a REGISTER request to at least two of the
>   default outbound proxies from the set. =E2=80=9C
>=20
> I will update the slides
An updated version (v1.1)  is now available at
http://www.slideshare.net/oej/sip-half-outbound-random-notes

/O
>>=20
>>=20
>> Q2: Flow-management
>>=20
>> Slide #14 says =C5=82No flow-management=CB=9B. What is meant by that? =
That the
>> keep-olives aren=C4=85t needed?
>>=20
> That=E2=80=99s very badly written by me. We need flow management, but =
not in
> the outbound-style, with flow tokens and all that. Will update.
>=20
> Thank you for very good feedback!
>=20
> /O
>>=20
>> Regards,
>>=20
>> Christer
>>=20
>>=20
>>=20
>> On 10/06/16 12:46, "sipcore on behalf of Olle E. Johansson"
>> <sipcore-bounces@ietf.org on behalf of oej@edvina.net> wrote:
>>=20
>>> Friends,
>>> I=C4=85ve tried to collect my thoughts about TLS and Outbound in a =
small
>>> brainstorm-presentation.
>>>=20
>>> http://www.slideshare.net/oej/sip-half-outbound-random-notes
>>>=20
>>> The chairs have alerted me to the fact that this may not be a =
SIPcore
>>> issue which I agree with, but I think this is the proper starting =
point
>>> for a general discussion between SIP people before we either drop =
the
>>> issue or drag it into dispatch. My apologies to the chairs for using =
this
>>> attractive forum :-)
>>>=20
>>> In short:
>>>=20
>>> - very few seems to implement full outbound
>>> - some parts of outbound seems used
>>> - which parts is not documented
>>> - TLS between UA and server has a clear problem if outbound is not =
used
>>>=20
>>> I am toying with an idea for a simple extension that puts the
>>> connection/flow management on the UA,
>>> allows the server to reuse the inbound TLS connection (without a =
client
>>> cert) after auth for outbound
>>> requests and also allows the server to remove associations in the
>>> location database at connection close.
>>>=20
>>> I think this documents what=C4=85s in use and formalises it with an =
extension
>>> tag.
>>>=20
>>> Please check my presentation and give me feedback.
>>>=20
>>> Thanks!
>>> /O
>>>=20
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>=20
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>=20


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From: "Drage, Keith (Nokia - GB)" <keith.drage@nokia.com>
To: "Olle E. Johansson" <oej@edvina.net>, SIPCORE <sipcore@ietf.org>
Thread-Topic: [sipcore] SIPS must die, die, die
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From nobody Fri Jun 10 13:00:16 2016
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] Half-outbound thoughts
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On 6/10/16 6:10 AM, Christer Holmberg wrote:
> Hi O,
>
> A couple of questions on the slides:
>
> Q1: Number of flows
>
> You claim that Outbound mandates at least two flows.
>
> I can find text saying that a UA must, as a minimum, support a set of two
> outbound proxy URIs, and while the whole idea of Outbound is having
> multiple flows I don¹t find any text saying that one must configure two
> (or more) outbound proxy URIs.

+1.

I just recently had occasion to review outbound for use in a specialized 
context, and also reached this conclusion. I think it can be used as-is 
for cases with only one flow.

The key is that you must *support* multiple flows, but the use of them 
is governed by the number of addresses that are configured. If only one 
is configured then you only have one flow.

	Thanks,
	Paul


From nobody Fri Jun 10 15:07:13 2016
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From: worley@ariadne.com (Dale R. Worley)
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Subject: Re: [sipcore] SIPS must die, die, die
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"Olle E. Johansson" <oej@edvina.net> writes:
> I think it's time to spend some time with TLS in SIPcore again. What
> do you think?

My memory is that the latest revision decided, as Keith says, that if
you make a call to a sips: URI, you are guaranteed that every signaling
step is covered by TLS, or the call is rejected -- and "transport=TLS"
doesn't have that guarantee.

Dale


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Dale R. Worley" <worley@ariadne.com>, "Olle E. Johansson" <oej@edvina.net>
Thread-Topic: [sipcore] SIPS must die, die, die
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Subject: Re: [sipcore] SIPS must die, die, die
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Hi,

>> I think it's time to spend some time with TLS in SIPcore again. What=20
>> do you think?
>
> My memory is that the latest revision decided, as Keith says, that if you=
 make a call to a sips: URI, you > are guaranteed that every signaling step=
 is covered by TLS, or the call is rejected

...or the SIPS URI is "converted" to a SIP URI by an intermediary. In reali=
ty there is no guarantee.

Regards,

Christer



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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Olle E. Johansson" <oej@edvina.net>
Thread-Topic: [sipcore] Half-outbound thoughts
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From: "Drage, Keith (Nokia - GB)" <keith.drage@nokia.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "Dale R. Worley" <worley@ariadne.com>, "Olle E. Johansson" <oej@edvina.net>
Thread-Topic: [sipcore] SIPS must die, die, die
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Date: Sat, 11 Jun 2016 13:53:44 +0000
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Subject: Re: [sipcore] SIPS must die, die, die
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The current status of the RFCs is clear.

While this behaviour may occur, it is non-conformant with the current requi=
rements of the RFCs, and IETF cannot be the protocol police.

Keith

-----Original Message-----
From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmb=
erg
Sent: 11 June 2016 07:48
To: Dale R. Worley; Olle E. Johansson
Cc: sipcore@ietf.org
Subject: Re: [sipcore] SIPS must die, die, die

Hi,

>> I think it's time to spend some time with TLS in SIPcore again. What=20
>> do you think?
>
> My memory is that the latest revision decided, as Keith says, that if=20
> you make a call to a sips: URI, you > are guaranteed that every=20
> signaling step is covered by TLS, or the call is rejected

...or the SIPS URI is "converted" to a SIP URI by an intermediary. In reali=
ty there is no guarantee.

Regards,

Christer


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


From nobody Sat Jun 11 07:11:51 2016
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Drage, Keith (Nokia - GB)" <keith.drage@nokia.com>, "Dale R. Worley" <worley@ariadne.com>, "Olle E. Johansson" <oej@edvina.net>
Thread-Topic: [sipcore] SIPS must die, die, die
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Subject: Re: [sipcore] SIPS must die, die, die
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--_000_7594FB04B1934943A5C02806D1A2204B3804BAA0ESESSMB209erics_
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I don't disagree.

Regards,

Christer

Sent from my Windows Phone
________________________________
From: Drage, Keith (Nokia - GB)<mailto:keith.drage@nokia.com>
Sent: =FD11/=FD06/=FD2016 16:53
To: Christer Holmberg<mailto:christer.holmberg@ericsson.com>; Dale R. Worle=
y<mailto:worley@ariadne.com>; Olle E. Johansson<mailto:oej@edvina.net>
Cc: sipcore@ietf.org<mailto:sipcore@ietf.org>
Subject: RE: [sipcore] SIPS must die, die, die

The current status of the RFCs is clear.

While this behaviour may occur, it is non-conformant with the current requi=
rements of the RFCs, and IETF cannot be the protocol police.

Keith

-----Original Message-----
From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmb=
erg
Sent: 11 June 2016 07:48
To: Dale R. Worley; Olle E. Johansson
Cc: sipcore@ietf.org
Subject: Re: [sipcore] SIPS must die, die, die

Hi,

>> I think it's time to spend some time with TLS in SIPcore again. What
>> do you think?
>
> My memory is that the latest revision decided, as Keith says, that if
> you make a call to a sips: URI, you > are guaranteed that every
> signaling step is covered by TLS, or the call is rejected

...or the SIPS URI is "converted" to a SIP URI by an intermediary. In reali=
ty there is no guarantee.

Regards,

Christer


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

--_000_7594FB04B1934943A5C02806D1A2204B3804BAA0ESESSMB209erics_
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<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dwindows-1=
256">
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
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<body>
<div>
<div>
<div style=3D"font-family:Calibri,sans-serif; font-size:11pt">I don't disag=
ree.<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:keith.drage@nokia.com">Drage, Keith (Nokia - GB)</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">=FD11=
/=FD06/=FD2016 16:53</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:christer.holmberg@ericsson.com">Christer Holmberg</a>;
<a href=3D"mailto:worley@ariadne.com">Dale R. Worley</a>; <a href=3D"mailto=
:oej@edvina.net">
Olle E. Johansson</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Cc:
</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] SIPS must die, die, die</span><br>
<br>
</div>
</div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">The current status of the RFCs is clear.<br>
<br>
While this behaviour may occur, it is non-conformant with the current requi=
rements of the RFCs, and IETF cannot be the protocol police.<br>
<br>
Keith<br>
<br>
-----Original Message-----<br>
From: sipcore [<a href=3D"mailto:sipcore-bounces@ietf.org">mailto:sipcore-b=
ounces@ietf.org</a>] On Behalf Of Christer Holmberg<br>
Sent: 11 June 2016 07:48<br>
To: Dale R. Worley; Olle E. Johansson<br>
Cc: sipcore@ietf.org<br>
Subject: Re: [sipcore] SIPS must die, die, die<br>
<br>
Hi,<br>
<br>
&gt;&gt; I think it's time to spend some time with TLS in SIPcore again. Wh=
at <br>
&gt;&gt; do you think?<br>
&gt;<br>
&gt; My memory is that the latest revision decided, as Keith says, that if =
<br>
&gt; you make a call to a sips: URI, you &gt; are guaranteed that every <br=
>
&gt; signaling step is covered by TLS, or the call is rejected<br>
<br>
...or the SIPS URI is &quot;converted&quot; to a SIP URI by an intermediary=
. In reality there is no guarantee.<br>
<br>
Regards,<br>
<br>
Christer<br>
<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>
</div>
</span></font>
</body>
</html>

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From: "DOLLY, MARTIN C" <md3135@att.com>
To: "Drage, Keith (Nokia - GB)" <keith.drage@nokia.com>
Thread-Topic: [sipcore] SIPS must die, die, die
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Subject: Re: [sipcore] SIPS must die, die, die
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God forbid real life implementation gets considered....   I could continue =
but considering the last discussion that shut down, I'll leave it at that

Martin C Dolly
Lead Member of Technical Staff
Core & Government/Regulatory Standards=20
AT&T
Cell: 609-903-3360
Email: md3135@att.com

> On Jun 11, 2016, at 9:54 AM, Drage, Keith (Nokia - GB) <keith.drage@nokia=
.com> wrote:
>=20
> The current status of the RFCs is clear.
>=20
> While this behaviour may occur, it is non-conformant with the current req=
uirements of the RFCs, and IETF cannot be the protocol police.
>=20
> Keith
>=20
> -----Original Message-----
> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Hol=
mberg
> Sent: 11 June 2016 07:48
> To: Dale R. Worley; Olle E. Johansson
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] SIPS must die, die, die
>=20
> Hi,
>=20
>>> I think it's time to spend some time with TLS in SIPcore again. What=20
>>> do you think?
>>=20
>> My memory is that the latest revision decided, as Keith says, that if=20
>> you make a call to a sips: URI, you > are guaranteed that every=20
>> signaling step is covered by TLS, or the call is rejected
>=20
> ...or the SIPS URI is "converted" to a SIP URI by an intermediary. In rea=
lity there is no guarantee.
>=20
> Regards,
>=20
> Christer
>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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From: "Drage, Keith (Nokia - GB)" <keith.drage@nokia.com>
To: "DOLLY, MARTIN C" <md3135@att.com>
Thread-Topic: [sipcore] SIPS must die, die, die
Thread-Index: AQHRw2RtEQdB19tCFkq/OxaVh7fUFZ/j0xwwgABl9+D///UYgIAAS7Fw
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Subject: Re: [sipcore] SIPS must die, die, die
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5630 was there to make a precise statement about what the use of SIPS URI v=
ersus transport=3Dtls meant. The former means the user wants TLS on every h=
op between the endpoints, the latter applies only to the current hop, with =
possibly some implication I trust my provider to do the rest.

Now you might argue there is space in there for "give me TLS on these bits,=
 and I'll trust the rest to be secure" but that would seem to be getting a =
bit complicated to define.

Otherwise I would argue that deployments should respect the requirements of=
 what a SIPS URI means, and only seek to improve on transport=3DTLS.

Keith

-----Original Message-----
From: DOLLY, MARTIN C [mailto:md3135@att.com]=20
Sent: 11 June 2016 15:12
To: Drage, Keith (Nokia - GB)
Cc: Christer Holmberg; Dale R. Worley; Olle E. Johansson; sipcore@ietf.org
Subject: Re: [sipcore] SIPS must die, die, die

God forbid real life implementation gets considered....   I could continue =
but considering the last discussion that shut down, I'll leave it at that

Martin C Dolly
Lead Member of Technical Staff
Core & Government/Regulatory Standards
AT&T
Cell: 609-903-3360
Email: md3135@att.com

> On Jun 11, 2016, at 9:54 AM, Drage, Keith (Nokia - GB) <keith.drage@nokia=
.com> wrote:
>=20
> The current status of the RFCs is clear.
>=20
> While this behaviour may occur, it is non-conformant with the current req=
uirements of the RFCs, and IETF cannot be the protocol police.
>=20
> Keith
>=20
> -----Original Message-----
> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer=20
> Holmberg
> Sent: 11 June 2016 07:48
> To: Dale R. Worley; Olle E. Johansson
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] SIPS must die, die, die
>=20
> Hi,
>=20
>>> I think it's time to spend some time with TLS in SIPcore again. What=20
>>> do you think?
>>=20
>> My memory is that the latest revision decided, as Keith says, that if=20
>> you make a call to a sips: URI, you > are guaranteed that every=20
>> signaling step is covered by TLS, or the call is rejected
>=20
> ...or the SIPS URI is "converted" to a SIP URI by an intermediary. In rea=
lity there is no guarantee.
>=20
> Regards,
>=20
> Christer
>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From nobody Sat Jun 11 09:59:53 2016
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] SIPS must die, die, die
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On 6/11/16 2:47 AM, Christer Holmberg wrote:
> Hi,
>
>>> I think it's time to spend some time with TLS in SIPcore again. What
>>> do you think?
>>
>> My memory is that the latest revision decided, as Keith says, that if you make a call to a sips: URI, you > are guaranteed that every signaling step is covered by TLS, or the call is rejected
>
> ...or the SIPS URI is "converted" to a SIP URI by an intermediary. In reality there is no guarantee.

So the choices are:

1) use SIPS as defined, meeting all the constraints
2) use SIPS but cheat when convenient
3) don't use SIPS

Can anyone cite are real deployments that meet (1)? I suspect there are 
none.

I hope nobody here is advocating (2). It has no useful security properties.

Hence ISTM that (3) is the only deployable alternative.

	Thanks,
	Paul


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From: "DOLLY, MARTIN C" <md3135@att.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [sipcore] SIPS must die, die, die
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Date: Sun, 12 Jun 2016 08:02:39 +0000
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Subject: Re: [sipcore] SIPS must die, die, die
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I vote 3, don't use SIPS

Martin C Dolly
Lead Member of Technical Staff
Core & Government/Regulatory Standards=20
AT&T
Cell: 609-903-3360
Email: md3135@att.com

> On Jun 11, 2016, at 1:00 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>=20
>> On 6/11/16 2:47 AM, Christer Holmberg wrote:
>> Hi,
>>=20
>>>> I think it's time to spend some time with TLS in SIPcore again. What
>>>> do you think?
>>>=20
>>> My memory is that the latest revision decided, as Keith says, that if y=
ou make a call to a sips: URI, you > are guaranteed that every signaling st=
ep is covered by TLS, or the call is rejected
>>=20
>> ...or the SIPS URI is "converted" to a SIP URI by an intermediary. In re=
ality there is no guarantee.
>=20
> So the choices are:
>=20
> 1) use SIPS as defined, meeting all the constraints
> 2) use SIPS but cheat when convenient
> 3) don't use SIPS
>=20
> Can anyone cite are real deployments that meet (1)? I suspect there are n=
one.
>=20
> I hope nobody here is advocating (2). It has no useful security propertie=
s.
>=20
> Hence ISTM that (3) is the only deployable alternative.
>=20
>    Thanks,
>    Paul
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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Subject: Re: [sipcore] Half-outbound thoughts
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> On 11 Jun 2016, at 08:54, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>=20
> Hi,
>=20
> Another question for clarification.
>=20
> Q3:
>=20
> Slide #2 talks about "outbound requests". What does "outbound" refer =
to? Requests somehow associated with the outbound mechanism, or the =
direction of the requests?
>=20
> If it's about the direction, shouldn't it then me "inbound", as the =
requests is from the server to the UA?
>=20
> If it's about requests associated with the outbound mechanism, I think =
you need to clarify a bit what exactly you mean by that.
Ok, I=E2=80=99ll take a look.

Thank you!
/O
>=20
> Regards,
>=20
> Christer
>=20
>=20
> -----Original Message-----
> From: Olle E. Johansson [mailto:oej@edvina.net]=20
> Sent: 10 June 2016 15:44
> To: Christer Holmberg <christer.holmberg@ericsson.com>
> Cc: Olle E Johansson <oej@edvina.net>; SIPCORE <sipcore@ietf.org>
> Subject: Re: [sipcore] Half-outbound thoughts
>=20
>=20
>> On 10 Jun 2016, at 13:04, Olle E. Johansson <oej@edvina.net> wrote:
>>=20
>>=20
>>> On 10 Jun 2016, at 12:10, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>>>=20
>>> Hi O,
>>>=20
>>> A couple of questions on the slides:
>>>=20
>>> Q1: Number of flows
>>>=20
>>> You claim that Outbound mandates at least two flows.
>>>=20
>>> I can find text saying that a UA must, as a minimum, support a set =
of=20
>>> two outbound proxy URIs, and while the whole idea of Outbound is=20
>>> having multiple flows I don=C4=85t find any text saying that one =
must=20
>>> configure two (or more) outbound proxy URIs.
>> We=E2=80=99ve discussed this many times at SIPit and have parsed TWO =
as a=20
>> must. I think you may be right though, that two is not required to=20
>> run, but the client must support using it. My failure.
>>=20
>> Additional quote:
>>=20
>> "If the set has more than
>>  one URI, the UAC MUST send a REGISTER request to at least two of the
>>  default outbound proxies from the set. =E2=80=9C
>>=20
>> I will update the slides
> An updated version (v1.1)  is now available at =
http://www.slideshare.net/oej/sip-half-outbound-random-notes
>=20
> /O
>>>=20
>>>=20
>>> Q2: Flow-management
>>>=20
>>> Slide #14 says =C5=82No flow-management=CB=9B. What is meant by =
that? That the=20
>>> keep-olives aren=C4=85t needed?
>>>=20
>> That=E2=80=99s very badly written by me. We need flow management, but =
not in=20
>> the outbound-style, with flow tokens and all that. Will update.
>>=20
>> Thank you for very good feedback!
>>=20
>> /O
>>>=20
>>> Regards,
>>>=20
>>> Christer
>>>=20
>>>=20
>>>=20
>>> On 10/06/16 12:46, "sipcore on behalf of Olle E. Johansson"
>>> <sipcore-bounces@ietf.org on behalf of oej@edvina.net> wrote:
>>>=20
>>>> Friends,
>>>> I=C4=85ve tried to collect my thoughts about TLS and Outbound in a =
small=20
>>>> brainstorm-presentation.
>>>>=20
>>>> http://www.slideshare.net/oej/sip-half-outbound-random-notes
>>>>=20
>>>> The chairs have alerted me to the fact that this may not be a=20
>>>> SIPcore issue which I agree with, but I think this is the proper=20
>>>> starting point for a general discussion between SIP people before =
we=20
>>>> either drop the issue or drag it into dispatch. My apologies to the=20=

>>>> chairs for using this attractive forum :-)
>>>>=20
>>>> In short:
>>>>=20
>>>> - very few seems to implement full outbound
>>>> - some parts of outbound seems used
>>>> - which parts is not documented
>>>> - TLS between UA and server has a clear problem if outbound is not=20=

>>>> used
>>>>=20
>>>> I am toying with an idea for a simple extension that puts the=20
>>>> connection/flow management on the UA, allows the server to reuse =
the=20
>>>> inbound TLS connection (without a client
>>>> cert) after auth for outbound
>>>> requests and also allows the server to remove associations in the=20=

>>>> location database at connection close.
>>>>=20
>>>> I think this documents what=C4=85s in use and formalises it with an=20=

>>>> extension tag.
>>>>=20
>>>> Please check my presentation and give me feedback.
>>>>=20
>>>> Thanks!
>>>> /O
>>>>=20
>>>> _______________________________________________
>>>> sipcore mailing list
>>>> sipcore@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>=20
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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Subject: Re: [sipcore] SIPS must die, die, die
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> On 10 Jun 2016, at 20:03, Drage, Keith (Nokia - GB) =
<keith.drage@nokia.com> wrote:
>=20
> I agree that the SIPS URI is not dead, but it is clearly stated in RFC =
5630 (which updates RFC 3261) that a successful call using a SIPS URI =
requires the use of TLS on each hop. Requests where this cannot be =
fulfilled have to be rejected.
>=20
> That is surely a big enough hint to only use it where that is what you =
want.
That=E2=80=99s the theory. But where=E2=80=99s the practise? I=E2=80=99ve =
seen old SIPura/linksys/cisco phones where you could add a prefix to the =
phone number and request a SIPS call. I guess zero calls was made using =
that feature

And even if the RFC says that TLS must be used in each hop, there=E2=80=99=
s no guarantee of anything, especially if the call is routed over =
another
technology somewhere, like ISDN. Is that even allowed for SIPS?

We struggled with this for a while in Asterisk, which is a multiprotocol =
b2bua and did not find any good solution on how to handle SIPS
in the dialplan or at all. And there was nothing to test with.

For me, I haven=E2=80=99t seen any use of SIPS anywhere. I do see a lot =
of TLS, used in default mode or opportunistic mode. But with SIP:=20
URI schemes.

What=E2=80=99s the use case here? Is anyone really using SIPS anywhere =
and is that use-case not possible to handle without SIPS: in the URI?

/O
>=20
> Keith
>=20
> -----Original Message-----
> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Olle E. =
Johansson
> Sent: 10 June 2016 10:52
> To: SIPCORE
> Cc: Olle E Johansson
> Subject: [sipcore] SIPS must die, die, die
>=20
> Friends,
>=20
> I think it=E2=80=99s time to kill the SIPS: uri. I know that some of =
you thought it was dead, but it is not the case.
>=20
> It doesn=E2=80=99t add anything, personally I don=E2=80=99t see it in =
use anywhere and it confuses a lot of discussions about TLS usage in =
SIP.
>=20
> We need to see what happens if we remove SIPS: from the SIP playground =
- what=E2=80=99s missing?
>=20
> Questions arise:
>=20
> - Can we live without SIPS: and without =E2=80=9C;transport=3DTLS=E2=80=9D=
 or do we need to take the transport parameter back?
> - Can my TLS connection reuse idea (previous mail) solve UA to edge =
proxy issues?
> - Do we need TLS transport in VIA headers?
> - Are we going to have to prepare for DTLS transport too? (See Dale =
Worley=E2=80=99s previous emails) ?
>=20
> I think it=E2=80=99s time to spend some time with TLS in SIPcore =
again. What do you think?
>=20
> /O
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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From: "Drage, Keith (Nokia - GB)" <keith.drage@nokia.com>
To: "DOLLY, MARTIN C" <md3135@att.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [sipcore] SIPS must die, die, die
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Subject: Re: [sipcore] SIPS must die, die, die
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That option was clearly understood by the SIP WG when we did RFC 5630. Howe=
ver it was decided to very clearly state what it meant so that both service=
 providers and users could decide whether to use it or not, and they still =
have that freedom. I suspect the majority have chosen not to, but the optio=
n is still there.

Given that we spent so much time on providing RFC 5630 to update RFC 3261 i=
n this area, I am reluctant to change the direction now - this achieved IET=
F consensus.

Keith

P.S. I also take this opportunity to remember Francois Audet, who did such =
a lot of work driving this document to completion.

-----Original Message-----
From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of DOLLY, MARTIN =
C
Sent: 12 June 2016 09:03
To: Paul Kyzivat
Cc: sipcore@ietf.org
Subject: Re: [sipcore] SIPS must die, die, die

I vote 3, don't use SIPS

Martin C Dolly
Lead Member of Technical Staff
Core & Government/Regulatory Standards
AT&T
Cell: 609-903-3360
Email: md3135@att.com

> On Jun 11, 2016, at 1:00 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>=20
>> On 6/11/16 2:47 AM, Christer Holmberg wrote:
>> Hi,
>>=20
>>>> I think it's time to spend some time with TLS in SIPcore again.=20
>>>> What do you think?
>>>=20
>>> My memory is that the latest revision decided, as Keith says, that=20
>>> if you make a call to a sips: URI, you > are guaranteed that every=20
>>> signaling step is covered by TLS, or the call is rejected
>>=20
>> ...or the SIPS URI is "converted" to a SIP URI by an intermediary. In re=
ality there is no guarantee.
>=20
> So the choices are:
>=20
> 1) use SIPS as defined, meeting all the constraints
> 2) use SIPS but cheat when convenient
> 3) don't use SIPS
>=20
> Can anyone cite are real deployments that meet (1)? I suspect there are n=
one.
>=20
> I hope nobody here is advocating (2). It has no useful security propertie=
s.
>=20
> Hence ISTM that (3) is the only deployable alternative.
>=20
>    Thanks,
>    Paul
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

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


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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "Drage, Keith (Nokia - GB)" <keith.drage@nokia.com>, "DOLLY, MARTIN C" <md3135@att.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [sipcore] SIPS must die, die, die
Thread-Index: AQHRw2RtJJyRZYItSEGGroXfIB854p/kFpuAgACq/ACAAPxBgIAB9SyA///VhwA=
Date: Mon, 13 Jun 2016 18:24:14 +0000
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Subject: Re: [sipcore] SIPS must die, die, die
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On the question of "cheating," the SIPS mechanism was originally intended
to be used in concert with DNS records that advertised SIPS as a service:
see the security considerations of RFC3263, for example. I'm not aware
that RFC5630 overwrote any of that guidance. In some scenarios, this could
make it possible to detect an intermediary downgrade from SIPS to SIP, as
both the endpoints - the UAC that originally targeted a SIPS URI and the
UAS or destination proxy advertising the URI - can collaborate to make
sure SIPS is requested and expected.

In highly mediated networks where SIP requests transit many hops, I'm sure
SIPS is at best a nuisance. Its design optimized more for a trapezoid, or
something close to it, than a PSTN-like network. I'd hesitate though to
eliminate a signaling confidentiality mode for SIP that has some
possibility of preventing pervasive metadata collection, even if we know
there are some environments where that protection will not be provided.
HTTPS is a nuisance in many highly mediated parts of the web, too.

Jon Peterson
Neustar, Inc.

On 6/13/16, 6:56 AM, "sipcore on behalf of Drage, Keith (Nokia - GB)"
<sipcore-bounces@ietf.org on behalf of keith.drage@nokia.com> wrote:

>That option was clearly understood by the SIP WG when we did RFC 5630.
>However it was decided to very clearly state what it meant so that both
>service providers and users could decide whether to use it or not, and
>they still have that freedom. I suspect the majority have chosen not to,
>but the option is still there.
>
>Given that we spent so much time on providing RFC 5630 to update RFC 3261
>in this area, I am reluctant to change the direction now - this achieved
>IETF consensus.
>
>Keith
>
>P.S. I also take this opportunity to remember Francois Audet, who did
>such a lot of work driving this document to completion.
>
>-----Original Message-----
>From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of DOLLY,
>MARTIN C
>Sent: 12 June 2016 09:03
>To: Paul Kyzivat
>Cc: sipcore@ietf.org
>Subject: Re: [sipcore] SIPS must die, die, die
>
>I vote 3, don't use SIPS
>
>Martin C Dolly
>Lead Member of Technical Staff
>Core & Government/Regulatory Standards
>AT&T
>Cell: 609-903-3360
>Email: md3135@att.com
>
>> On Jun 11, 2016, at 1:00 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>>=20
>>> On 6/11/16 2:47 AM, Christer Holmberg wrote:
>>> Hi,
>>>=20
>>>>> I think it's time to spend some time with TLS in SIPcore again.
>>>>> What do you think?
>>>>=20
>>>> My memory is that the latest revision decided, as Keith says, that
>>>> if you make a call to a sips: URI, you > are guaranteed that every
>>>> signaling step is covered by TLS, or the call is rejected
>>>=20
>>> ...or the SIPS URI is "converted" to a SIP URI by an intermediary. In
>>>reality there is no guarantee.
>>=20
>> So the choices are:
>>=20
>> 1) use SIPS as defined, meeting all the constraints
>> 2) use SIPS but cheat when convenient
>> 3) don't use SIPS
>>=20
>> Can anyone cite are real deployments that meet (1)? I suspect there are
>>none.
>>=20
>> I hope nobody here is advocating (2). It has no useful security
>>properties.
>>=20
>> Hence ISTM that (3) is the only deployable alternative.
>>=20
>>    Thanks,
>>    Paul
>>=20
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>
>_______________________________________________
>sipcore mailing list
>sipcore@ietf.org
>https://www.ietf.org/mailman/listinfo/sipcore
>
>_______________________________________________
>sipcore mailing list
>sipcore@ietf.org
>https://www.ietf.org/mailman/listinfo/sipcore


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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Thread-Topic: [sipcore] SIP AuthNZ Problem Statement - v3
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Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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--_000_D3842CA219BA97jonpetersonneustarbiz_
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Just to play back the general problem statement, I think we'd agree that of=
 your three numbered possible solutions, #1 is more or less the assumed bas=
eline SIP behavior - cases where, for example, the server would send a Dige=
st challenge in the backwards direction to authenticate the user agent and =
then make any authorization decision based on its own ACLs or what have you=
. Your solutions #2 and #3 cover the two cases I think are interesting: SSO=
-like ones where relying parties trust an identity assertion provided by an=
 IdP of some sort (#2) and cases where the identity information is not incl=
uded at all (or is at least optional) and instead the relying party trusts =
traits or attributes or what have you provided by the IdP. I've argued that=
 SIP has always been able to do #1, that we at least already have some ways=
 to do #2, and that we've already done requirements for #3 (RFC4484) but to=
 date have gone no farther, and that it would be new work if we were to spe=
cify protocol for #3.

So yes, the third option is where we need to focus. But the point of my las=
t mail was to say that what we need to narrow our discussion down to what t=
he attributes are that we imagine applications in these use cases will requ=
ire to process SIP requests, primarily SIP INVITE requests. You say some hi=
gh-level things below like that your third-option token "might grant the us=
er access to multiple services" on the conference service which is your rel=
ying party. In terms of concrete authorization traits for this use case, to=
wards the very end you suggest attributes of a conference like the number o=
f participants or the media types permitted in the conference.

Radhika sent a helpful summary of conference control protocols to the list =
which surveyed mechanisms like CCMP, BFCP, and MRB, protocols which actuall=
y let you manage attributes of a conference like the number of participants=
 and media types allowed. The interesting thing about protocols like CCMP i=
s that they are not, you know, actually transported by SIP. CCMP is transpo=
rted by HTTP. Why? Because the semantics of conference control doesn't actu=
ally map onto SIP INVITE transactions very well. They map quite aptly to HT=
TP, and in the HTTP context, you can certainly use OAuth if you like. You e=
ven will find some text about Role-Based Access Control in the XCON (RFC650=
3) security consideration. Although you can use SIP and SDP to set up a BFC=
P connection (RFC4583), the security of BFCP relies on mutual TLS between i=
ts own endpoints. It is unclear to me what carrying attribute information a=
round in a SIP INVITE would accomplish with regards to authorizing CCMP or =
BFCP transactions.

So again, the point of my last mail was that the problem of authorizing SIP=
 INVITEs is actually different from the problem of authorizing generic appl=
ication transactions. SIP INVITEs still have more or less the semantics of =
a call setup message: they are requesting that a session be initiated. Conf=
erence control is not managed with SIP transactions, and the security of co=
nference control protocols does not devolve to authorizing SIP transactions=
. I suspect we'll find that most of the other security attributes that "app=
lications" need to do their jobs similarly don't map onto the semantics of =
a SIP INVITE - but, I'm still open to entertaining examples that might cast=
 doubt on that suspicion.

Jon Peterson
Neustar, Inc.

From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com<mailto:rifaat.ietf@gmail.co=
m>>
Date: Thursday, June 9, 2016 at 10:59 AM
To: Jon Peterson <jon.peterson@neustar.biz<mailto:jon.peterson@neustar.biz>=
>
Cc: Christer Holmberg <christer.holmberg@ericsson.com<mailto:christer.holmb=
erg@ericsson.com>>, "sipcore@ietf.org<mailto:sipcore@ietf.org>" <sipcore@ie=
tf.org<mailto:sipcore@ietf.org>>
Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3

Hi Jon,

Sorry for the delayed reply.

Let's take the video conference example, and discuss the possible ways for =
the conference server to provide service.


         Trust Domain 1              :    Internet     :  Trust Domain 2
                                     :                 :
                      +--------+     :                 :
                      | IdP    |     :                 :
        +-------------| AuthZ  |     :                 :
        |             |        |     :                 :
        |             +--------+     :                 :
        |                 |          :                 :
        |                 |          :                 :
    +--------+        +--------+     :                 :     +--------+
    |        |        | SIP    |     :                 :     | SIP App|
    |   UA   |--------| Proxy  |-----:-----------------:-----| Video  |
    |        |        |        |     :                 :     | Conference
    +--------+        +--------+     :                 :     +--------+
                                     :                 :

Possible solutions:

1. Conference Server acts as an AuthN and AuthZ server. Requests to the ser=
ver must provide an identity and credentials, and services will be provided=
 based on locally stored information associated with the identity provided =
in the request.

2. Conference Server acts as an AuthZ server, but delegates the AuthN part.=
 Requests to the server must provide an identity token, and services will b=
e provided based on locally stored information associated with the identity=
 provided in the token.

3. Conference Server delegates both AuthN and AuthZ parts. Requests to the =
server must provide an access token, which might or might not include infor=
mation about the user requesting the service, and services will be provided=
 based on information provided in the access token or as a result of an int=
rospection step using the access token.


Now, let's look at the 3rd option a bit more, and assume that the server pr=
ovides the following types of conference services:
1. audio conference
2. audio/video conference
3. web conference

or some combination of the above.

Let's also assume that the server allows control over the number of partici=
pants allowed in each one of these conference types or combinations.


Here is one way that I envisioned we could use to address this use case:

When an enterprise user in the 1st trust domain wants to register with the =
SIP Proxy, it will have to authenticate to the IdP and as a result either t=
he UA or the Proxy (depends on the UA's capabilities) will be provided with=
 an AuthN and AuthZ tokens. This process could involve an interim step to a=
llow only the proxy to get access to the tokens, or the tokens could be sen=
t to the UA which will then use them to communicate with the proxy.

The AuthN token is the identity of the user, and the AuthZ token has all th=
e services granted to the user by the IdP/AuthZ Service.

When the enterprise user needs access to the conference service, the UA wil=
l send an INVITE to the Proxy, and will include the AuthZ token if it is in=
 possession of the tokens; otherwise, the proxy will add the appropriate to=
ken to the outgoing INVITE sent to the conference server.

The scope of the AuthZ token might grant the user access to multiple servic=
es; to avoid sending that AuthZ token to all servers providing services gra=
nted to the user, the proxy could obtain a different token with limited sco=
pe that is specific to the service being provided (conference services in t=
his case); there is a standard Token Exchange mechanism for that.

The AuthZ token will include information about the type of service allowed =
(audio, audio/video, etc), and the limit on the number of participants. The=
 AuthZ token might or might not have any information about the user request=
ing the service, which depends on the level of privacy needed.

When the conference server receives this token in the SIP INVITE, all it ha=
s to do is validate that the token was signed by an IdP that it has a trust=
 relationship with, and provide the service requested by the AuthZ token. T=
he conference server does not need the AuthN token to be able to provide a =
service.

Regards,
 Rifaat


On Wed, May 25, 2016 at 3:06 PM, Peterson, Jon <jon.peterson@neustar.biz<ma=
ilto:jon.peterson@neustar.biz>> wrote:

Also, if we decided to separate the AuthN and AuthZ, then we might need to =
carry more than one token.

I could see an argument that there is a need to carry the identity of the u=
ser in one header, and a reference to a place where you can get attributes =
about the originating user in another header, say. That's more or less what=
 we envisioned the last time we went down this path, which led to the RFC44=
84 requirements, largely informed by SAML (which I seem to recall 3GPP was =
interested in at the time).

But it's no accident that the work on RFC4484 never went beyond requirement=
s. Today, I think that much of the useful work that could be done on the au=
thorization question is in identifying which SIP authorization policies are=
 likely to depend on attributes as opposed to just the identity of the end =
user. While we can readily conceive of applications adjunct to communicatio=
ns that would require attributes, the challenge has been justifying why you=
 would invoke those applications with SIP rather than, say, HTTP.  The fund=
amental semantics of a SIP INVITE have remained pretty close to those of pl=
acing a telephone call.  The authorization policies of a telephone call are=
 really different than that of a web service, say, both in terms of how and=
 why they are invoked, what kinds of parties are involved, what forms redir=
ection or cross-domain interaction can occur. With appropriate caveats for =
other methods (notably MESSAGE, SUBSCRIBE/NOTIFY. INFO), SIP requests have =
authorization policy requirements that look a lot like call treatment polic=
ies.

That much said, I am kind of interested in call treatment policies, and the=
ir interaction with identity information. But just saying that there might =
be a video conference service in the cloud, say, doesn't shed a lot of ligh=
t for me. If there's a video conference, I'd ordinarily think you just call=
 into it. If the user who opens the conference bridge needs to be charged, =
that's identity information - which in its usual form will convey the indiv=
idual user and their domain, like user@domain. If you need more complex sem=
antics than that, it's probably because you're doing something more complic=
ated than just calling in to the conference. But it's less clear to me why =
you'd use SIP for such cases. Drilling down deeper into these use cases wou=
ld help me understand where we're going, anyway - the use cases in RFC4484 =
Section 4, which largely involve billing (including when dealing with vario=
us protocol gateways) and resource management (both preemption and priority=
), ultimately didn't seem sufficient to motivate protocol work.

Another issues in the need, in some cases, to avoid sending the token to th=
e UA; in this case there is a need for an interim step to get the token(s) =
to the SIP proxy.

If this information is carried in headers, then presumably it could be adde=
d, consumed, subtracted, or whatever by any relevant intermediaries.

Jon Peterson
Neustar, Inc.


--_000_D3842CA219BA97jonpetersonneustarbiz_
Content-Type: text/html; charset="us-ascii"
Content-ID: <E612C7D20F3A134E8301F104872B0A56@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>Just to play back the general problem statement, I think we'd agree th=
at of your three numbered possible solutions, #1 is more or less the assume=
d baseline SIP behavior - cases where, for example, the server would send a=
 Digest challenge in the backwards
 direction to authenticate the user agent and then make any authorization d=
ecision based on its own ACLs or what have you. Your solutions #2 and #3 co=
ver the two cases I think are interesting: SSO-like ones where relying part=
ies trust an identity assertion
 provided by an IdP of some sort (#2) and cases where the identity informat=
ion is not included at all (or is at least optional) and instead the relyin=
g party trusts traits or attributes or what have you provided by the IdP. I=
've argued that SIP has always been
 able to do #1, that we at least already have some ways to do #2, and that =
we've already done requirements for #3 (RFC4484) but to date have gone no f=
arther, and that it would be new work if we were to specify protocol for #3=
.</div>
<div><br>
</div>
<div>So yes, the third option is where we need to focus. But the point of m=
y last mail was to say that what we need to narrow our discussion down to w=
hat the attributes are that we imagine applications in these use cases will=
 require to process SIP requests,
 primarily SIP INVITE requests. You say some high-level things below like t=
hat your third-option token &quot;might grant the user access to multiple s=
ervices&quot; on the conference service which is your relying party. In ter=
ms of concrete authorization traits for this
 use case, towards the very end you suggest attributes of a conference like=
 the number of participants or the media types permitted in the conference.=
</div>
<div><br>
</div>
<div>Radhika sent a helpful summary of conference control protocols to the =
list which surveyed mechanisms like CCMP, BFCP, and MRB, protocols which ac=
tually let you manage attributes of a conference like the number of partici=
pants and media types allowed. The
 interesting thing about protocols like CCMP is that they are not, you know=
, actually transported by SIP. CCMP is transported by HTTP. Why? Because th=
e semantics of conference control doesn't actually map onto SIP INVITE tran=
sactions very well. They map quite
 aptly to HTTP, and in the HTTP context, you can certainly use OAuth if you=
 like. You even will find some text about Role-Based Access Control in the =
XCON (RFC6503) security consideration. Although you can use SIP and SDP to =
set up a BFCP connection (RFC4583),
 the security of BFCP relies on mutual TLS between its own endpoints. It is=
 unclear to me what carrying attribute information around in a SIP INVITE w=
ould accomplish with regards to authorizing CCMP or BFCP transactions.</div=
>
<div><br>
</div>
<div>So again, the point of my last mail was that the problem of authorizin=
g SIP INVITEs is actually different from the problem of authorizing generic=
 application transactions. SIP INVITEs still have more or less the semantic=
s of a call setup message: they
 are requesting that a session be initiated. Conference control is not mana=
ged with SIP transactions, and the security of conference control protocols=
 does not devolve to authorizing SIP transactions. I suspect we'll find tha=
t most of the other security attributes
 that &quot;applications&quot; need to do their jobs similarly don't map on=
to the semantics of a SIP INVITE - but, I'm still open to entertaining exam=
ples that might cast doubt on that suspicion.&nbsp;</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>Thursday, June 9, 2016 at 10:=
59 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>Christer Holmberg &lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson.com</=
a>&gt;, &quot;<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a>&quot=
; &lt;<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [sipcore] SIP AuthNZ P=
roblem Statement - v3<br>
</div>
<div><br>
</div>
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div>
<div>
<div dir=3D"ltr"><font face=3D"monospace,monospace">Hi Jon,</font>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Sorry for the delayed reply.</font>=
</div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div>
<div>
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><font face=3D"monospace,monospace">Let's take the video conference exa=
mple, and discuss the possible ways for&nbsp;</font><span style=3D"font-fam=
ily:monospace,monospace">the conference server to provide service.</span></=
div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;T=
rust Domain 1 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;: &nbsp; &nbs=
p;Internet &nbsp; &nbsp; : &nbsp;Trust Domain 2</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &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; :</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &#43;--------&#43; &nbsp; &nbsp; =
: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | IdP &nbsp; &nbsp;| &nbsp; &nbsp=
; : &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; &#43;--=
-----------| AuthZ &nbsp;| &nbsp; &nbsp; : &nbsp; &nbsp; &nbsp; &nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; :</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; | &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; =
&nbsp; : &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</font></=
div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; | &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &#43;--------&#43; &nbsp; &nbsp; : &nb=
sp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; | &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &=
nbsp; &nbsp;: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</fo=
nt></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; | &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &=
nbsp; &nbsp;: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</fo=
nt></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &#43;--------&#43; &n=
bsp; &nbsp; &nbsp; &nbsp;&#43;--------&#43; &nbsp; &nbsp; : &nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : &nbsp; &nbsp; &#43;--------&#43;=
</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; | &nbsp; &nbsp; &nbsp=
; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp;| SIP &nbsp; &nbsp;| &nbsp; &nbsp; : &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : &nbsp; &nbsp; | SI=
P App|</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; | &nbsp; UA &nbsp; |-=
-------| Proxy &nbsp;|-----:-----------------:-----| Video &nbsp;|</font></=
div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; | &nbsp; &nbsp; &nbsp=
; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; =
&nbsp; : &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : &nbsp; &=
nbsp; | Conference</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &#43;--------&#43; &n=
bsp; &nbsp; &nbsp; &nbsp;&#43;--------&#43; &nbsp; &nbsp; : &nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : &nbsp; &nbsp; &#43;--------&#43;=
</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &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; :</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Possible solutions:</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">1. Conference Server acts as an Aut=
hN and AuthZ server. Requests to the&nbsp;server must provide an identity a=
nd credentials, and services will be&nbsp;provided based on locally stored =
information associated with the&nbsp;identity provided
 in the request.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">2. Conference Server acts as an Aut=
hZ server, but delegates the AuthN part.&nbsp;Requests to the server must p=
rovide an identity token, and services will&nbsp;be provided based on local=
ly stored information associated with the&nbsp;identity
 provided in the token.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">3. Conference Server delegates both=
 AuthN and AuthZ parts. Requests to the&nbsp;server must provide an access =
token, which might or might not include&nbsp;information about the user req=
uesting the service, and services will&nbsp;be provided
 based on information provided in the access token or as a&nbsp;result of a=
n introspection step using the access token.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Now, let's look at the 3rd option a=
 bit more, and assume that the server&nbsp;provides the following types of =
conference services:</font></div>
<div><font face=3D"monospace,monospace">1. audio conference</font></div>
<div><font face=3D"monospace,monospace">2. audio/video conference</font></d=
iv>
<div><font face=3D"monospace,monospace">3. web conference</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">or some&nbsp;combination&nbsp;of th=
e above.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Let's also assume that the server a=
llows control over the number of&nbsp;participants allowed in each one of t=
hese conference types or combinations.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Here is one way that I envisioned w=
e could use to address this use case:</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
</div>
</div>
</div>
<font face=3D"monospace,monospace">When an enterprise user in the 1st trust=
 domain wants to register with the&nbsp;SIP Proxy, it will have to authenti=
cate to the IdP and as a result either&nbsp;the UA or the Proxy (depends on=
 the UA's capabilities) will be provided&nbsp;with
 an AuthN and AuthZ tokens. This process could involve an interim step&nbsp=
;to allow only the proxy to get access to the tokens, or the tokens could b=
e&nbsp;sent to the UA which will then use them to communicate with the prox=
y.</font>
<div><font face=3D"monospace,monospace"><br>
The AuthN token is the identity of the user, and the AuthZ token has all&nb=
sp;the services granted to the user by the IdP/AuthZ Service.<br>
<br>
When the enterprise user needs access to the conference service, the UA&nbs=
p;will send an INVITE to the Proxy, and will include the AuthZ token if it =
is&nbsp;in possession of the tokens; otherwise, the proxy will add the appr=
opriate&nbsp;token to the outgoing INVITE sent
 to the conference server.<br>
<br>
The scope of the AuthZ token might grant the user access to multiple&nbsp;s=
ervices; to avoid sending that AuthZ token to all servers providing&nbsp;se=
rvices granted to the user, the proxy could obtain a different token with&n=
bsp;limited scope that is specific to the service
 being provided (conference&nbsp;services in this case); there is a standar=
d Token Exchange mechanism for&nbsp;that.<br>
<br>
The AuthZ token will include information about the type of service allowed&=
nbsp;(audio, audio/video, etc), and the limit on the number of participants=
.&nbsp;The AuthZ token might or might not have any information about the us=
er&nbsp;requesting the service, which depends on
 the level of privacy needed.<br>
<br>
When the conference server receives this token in the SIP INVITE, all it&nb=
sp;has to do is validate that the token was signed by an IdP that it has a&=
nbsp;trust relationship with, and provide the service requested by the Auth=
Z&nbsp;token. The conference server does not need
 the AuthN token to be able to&nbsp;provide a service.</font>
<div>
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Regards,</font></div>
<div><font face=3D"monospace,monospace">&nbsp;Rifaat</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
</div>
</div>
</div>
</div>
</div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Wed, May 25, 2016 at 3:06 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:0 0 0 .8ex;border-left:1p=
x #ccc 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 class=3D""><span>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
</div>
</div>
</div>
</span><span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>Also, if we decided to separate the AuthN and AuthZ, then we might nee=
d to carry more than one token.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span>
<div>I could see an argument that there is a need to carry the identity of =
the user in one header, and a reference to a place where you can get attrib=
utes about the originating user in another header, say. That's more or less=
 what we envisioned the last time
 we went down this path, which led to the RFC4484 requirements, largely inf=
ormed by SAML (which I seem to recall 3GPP was interested in at the time).<=
/div>
<div><br>
</div>
<div>But it's no accident that the work on RFC4484 never went beyond requir=
ements. Today, I think that much of the useful work that could be done on t=
he authorization question is in identifying which SIP authorization policie=
s are likely to depend on attributes
 as opposed to just the identity of the end user. While we can readily conc=
eive of applications adjunct to communications that would require attribute=
s, the challenge has been justifying why you would invoke those application=
s with SIP rather than, say, HTTP.
 &nbsp;The fundamental semantics of a SIP INVITE have remained pretty close=
 to those of placing a telephone call.&nbsp; The authorization policies of =
a telephone call are really different than that of a web service, say, both=
 in terms of how and why they are invoked,
 what kinds of parties are involved, what forms redirection or cross-domain=
 interaction can occur. With appropriate caveats for other methods (notably=
 MESSAGE, SUBSCRIBE/NOTIFY. INFO), SIP requests have authorization policy r=
equirements that look a lot like
 call treatment policies.</div>
<div><br>
</div>
<div>That much said, I am kind of interested in call treatment policies, an=
d their interaction with identity information. But just saying that there m=
ight be a video conference service in the cloud, say, doesn't shed a lot of=
 light for me. If there's a video
 conference, I'd ordinarily think you just call into it. If the user who op=
ens the conference bridge needs to be charged, that's identity information =
- which in its usual form will convey the individual user and their domain,=
 like user@domain. If you need more
 complex semantics than that, it's probably because you're doing something =
more complicated than just calling in to the conference. But it's less clea=
r to me why you'd use SIP for such cases. Drilling down deeper into these u=
se cases would help me understand
 where we're going, anyway - the use cases in RFC4484 Section 4, which larg=
ely involve billing (including when dealing with various protocol gateways)=
 and resource management (both preemption and priority), ultimately didn't =
seem sufficient to motivate protocol
 work.</div>
<span class=3D"">
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>Another issues in the need, in some cases, to avoid sending the token =
to the UA; in this case there is a need for an interim step to get the toke=
n(s) to the SIP proxy.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span>
<div>If this information is carried in headers, then presumably it could be=
 added, consumed, subtracted, or whatever by any relevant intermediaries.&n=
bsp;</div>
<span class=3D"HOEnZb"><font color=3D"#888888">
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
</font></span></div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</blockquote>
</span>
</body>
</html>

--_000_D3842CA219BA97jonpetersonneustarbiz_--


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From: Richard Shockey <richard@shockey.us>
To: "Peterson, Jon" <jon.peterson@neustar.biz>, "Drage, Keith (Nokia - GB)" <keith.drage@nokia.com>, "DOLLY, MARTIN C" <md3135@att.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>
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Thread-Topic: [sipcore] SIPS must die, die, die
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Subject: Re: [sipcore] SIPS must die, die, die
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On 6/13/16, 2:24 PM, "sipcore on behalf of Peterson, Jon" <sipcore-bounces@=
ietf.org on behalf of jon.peterson@neustar.biz> wrote:

>
>On the question of "cheating," the SIPS mechanism was originally intended
>to be used in concert with DNS records that advertised SIPS as a service:
>see the security considerations of RFC3263, for example. I'm not aware
>that RFC5630 overwrote any of that guidance. In some scenarios, this could
>make it possible to detect an intermediary downgrade from SIPS to SIP, as
>both the endpoints - the UAC that originally targeted a SIPS URI and the
>UAS or destination proxy advertising the URI - can collaborate to make
>sure SIPS is requested and expected.
>
>In highly mediated networks where SIP requests transit many hops, I'm sure
>SIPS is at best a nuisance.=20

RS> That is an understatement.  SIPS is not used in virtually any component=
 of the carrier based Public SIP Networks that rely on E.164 naming that I k=
now of. Though it is specified as MUST support in the SIP Forum SIPConnect p=
rofiles for PBX to SP on the wire integration to my knowledge it has never b=
een implemented by any vendor or SSP.  In short SIPS is a total Pain in the =
Ass.  Given the appalling track record the IETF has in designing implementab=
le security protocols, I question whether this effort is worth the time.  Bu=
t that=E2=80=99s OK ..if people want to tilt at windmills, be my guest.  What I wa=
nt to know is does anyone really care?=20


Its design optimized more for a trapezoid, or
>something close to it, than a PSTN-like network. I'd hesitate though to
>eliminate a signaling confidentiality mode for SIP that has some
>possibility of preventing pervasive metadata collection, even if we know
>there are some environments where that protection will not be provided.
>HTTPS is a nuisance in many highly mediated parts of the web, too.
>
>Jon Peterson
>Neustar, Inc.
>
>On 6/13/16, 6:56 AM, "sipcore on behalf of Drage, Keith (Nokia - GB)"
><sipcore-bounces@ietf.org on behalf of keith.drage@nokia.com> wrote:
>
>>That option was clearly understood by the SIP WG when we did RFC 5630.
>>However it was decided to very clearly state what it meant so that both
>>service providers and users could decide whether to use it or not, and
>>they still have that freedom. I suspect the majority have chosen not to,
>>but the option is still there.
>>
>>Given that we spent so much time on providing RFC 5630 to update RFC 3261
>>in this area, I am reluctant to change the direction now - this achieved
>>IETF consensus.
>>
>>Keith
>>
>>P.S. I also take this opportunity to remember Francois Audet, who did
>>such a lot of work driving this document to completion.
>>
>>-----Original Message-----
>>From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of DOLLY,
>>MARTIN C
>>Sent: 12 June 2016 09:03
>>To: Paul Kyzivat
>>Cc: sipcore@ietf.org
>>Subject: Re: [sipcore] SIPS must die, die, die
>>
>>I vote 3, don't use SIPS
>>
>>Martin C Dolly
>>Lead Member of Technical Staff
>>Core & Government/Regulatory Standards
>>AT&T
>>Cell: 609-903-3360
>>Email: md3135@att.com
>>
>>> On Jun 11, 2016, at 1:00 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote=
:
>>>=20
>>>> On 6/11/16 2:47 AM, Christer Holmberg wrote:
>>>> Hi,
>>>>=20
>>>>>> I think it's time to spend some time with TLS in SIPcore again.
>>>>>> What do you think?
>>>>>=20
>>>>> My memory is that the latest revision decided, as Keith says, that
>>>>> if you make a call to a sips: URI, you > are guaranteed that every
>>>>> signaling step is covered by TLS, or the call is rejected
>>>>=20
>>>> ...or the SIPS URI is "converted" to a SIP URI by an intermediary. In
>>>>reality there is no guarantee.
>>>=20
>>> So the choices are:
>>>=20
>>> 1) use SIPS as defined, meeting all the constraints
>>> 2) use SIPS but cheat when convenient
>>> 3) don't use SIPS
>>>=20
>>> Can anyone cite are real deployments that meet (1)? I suspect there are
>>>none.
>>>=20
>>> I hope nobody here is advocating (2). It has no useful security
>>>properties.
>>>=20
>>> Hence ISTM that (3) is the only deployable alternative.
>>>=20
>>>    Thanks,
>>>    Paul
>>>=20
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>
>>_______________________________________________
>>sipcore mailing list
>>sipcore@ietf.org
>>https://www.ietf.org/mailman/listinfo/sipcore
>>
>>_______________________________________________
>>sipcore mailing list
>>sipcore@ietf.org
>>https://www.ietf.org/mailman/listinfo/sipcore
>
>_______________________________________________
>sipcore mailing list
>sipcore@ietf.org
>https://www.ietf.org/mailman/listinfo/sipcore



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Subject: Re: [sipcore] SIPS must die, die, die
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> On 13 Jun 2016, at 20:24, Peterson, Jon <jon.peterson@neustar.biz> =
wrote:
>=20
>=20
> On the question of "cheating," the SIPS mechanism was originally =
intended
> to be used in concert with DNS records that advertised SIPS as a =
service:
> see the security considerations of RFC3263, for example. I'm not aware
> that RFC5630 overwrote any of that guidance. In some scenarios, this =
could
> make it possible to detect an intermediary downgrade from SIPS to SIP, =
as
> both the endpoints - the UAC that originally targeted a SIPS URI and =
the
> UAS or destination proxy advertising the URI - can collaborate to make
> sure SIPS is requested and expected.
>=20
> In highly mediated networks where SIP requests transit many hops, I'm =
sure
> SIPS is at best a nuisance. Its design optimized more for a trapezoid, =
or
> something close to it, than a PSTN-like network. I'd hesitate though =
to
> eliminate a signaling confidentiality mode for SIP that has some
> possibility of preventing pervasive metadata collection, even if we =
know
> there are some environments where that protection will not be =
provided.
> HTTPS is a nuisance in many highly mediated parts of the web, too.
>=20
I think we need to start with =E2=80=9Cwhat if we remove SIPS=E2=80=9D =
and go back and
see what=E2=80=99s missing and how we can fix that. A lot of people =
point to SIPS as
the solution - but no one is implementing or using it.=20

What is the problem it solves? Is there another way? What=E2=80=99s the =
status for=20
S/MIME.

I don=E2=80=99t think we=E2=80=99ll find that the answer is SIPS: as a =
URI scheme, but the
SIP over TLS transport is still very much needed, discovered by =
NAPTR/SRV
and supported by DANE or some other PKI structure.

/O

> Jon Peterson
> Neustar, Inc.
>=20
> On 6/13/16, 6:56 AM, "sipcore on behalf of Drage, Keith (Nokia - GB)"
> <sipcore-bounces@ietf.org on behalf of keith.drage@nokia.com> wrote:
>=20
>> That option was clearly understood by the SIP WG when we did RFC =
5630.
>> However it was decided to very clearly state what it meant so that =
both
>> service providers and users could decide whether to use it or not, =
and
>> they still have that freedom. I suspect the majority have chosen not =
to,
>> but the option is still there.
>>=20
>> Given that we spent so much time on providing RFC 5630 to update RFC =
3261
>> in this area, I am reluctant to change the direction now - this =
achieved
>> IETF consensus.
>>=20
>> Keith
>>=20
>> P.S. I also take this opportunity to remember Francois Audet, who did
>> such a lot of work driving this document to completion.
>>=20
>> -----Original Message-----
>> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of DOLLY,
>> MARTIN C
>> Sent: 12 June 2016 09:03
>> To: Paul Kyzivat
>> Cc: sipcore@ietf.org
>> Subject: Re: [sipcore] SIPS must die, die, die
>>=20
>> I vote 3, don't use SIPS
>>=20
>> Martin C Dolly
>> Lead Member of Technical Staff
>> Core & Government/Regulatory Standards
>> AT&T
>> Cell: 609-903-3360
>> Email: md3135@att.com
>>=20
>>> On Jun 11, 2016, at 1:00 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> =
wrote:
>>>=20
>>>> On 6/11/16 2:47 AM, Christer Holmberg wrote:
>>>> Hi,
>>>>=20
>>>>>> I think it's time to spend some time with TLS in SIPcore again.
>>>>>> What do you think?
>>>>>=20
>>>>> My memory is that the latest revision decided, as Keith says, that
>>>>> if you make a call to a sips: URI, you > are guaranteed that every
>>>>> signaling step is covered by TLS, or the call is rejected
>>>>=20
>>>> ...or the SIPS URI is "converted" to a SIP URI by an intermediary. =
In
>>>> reality there is no guarantee.
>>>=20
>>> So the choices are:
>>>=20
>>> 1) use SIPS as defined, meeting all the constraints
>>> 2) use SIPS but cheat when convenient
>>> 3) don't use SIPS
>>>=20
>>> Can anyone cite are real deployments that meet (1)? I suspect there =
are
>>> none.
>>>=20
>>> I hope nobody here is advocating (2). It has no useful security
>>> properties.
>>>=20
>>> Hence ISTM that (3) is the only deployable alternative.
>>>=20
>>>   Thanks,
>>>   Paul
>>>=20
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>=20
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>=20
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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Subject: Re: [sipcore] SIPS must die, die, die
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> On 13 Jun 2016, at 22:23, Richard Shockey <richard@shockey.us> wrote:
>=20
>=20
> On 6/13/16, 2:24 PM, "sipcore on behalf of Peterson, Jon" =
<sipcore-bounces@ietf.org on behalf of jon.peterson@neustar.biz> wrote:
>=20
>>=20
>> On the question of "cheating," the SIPS mechanism was originally =
intended
>> to be used in concert with DNS records that advertised SIPS as a =
service:
>> see the security considerations of RFC3263, for example. I'm not =
aware
>> that RFC5630 overwrote any of that guidance. In some scenarios, this =
could
>> make it possible to detect an intermediary downgrade from SIPS to =
SIP, as
>> both the endpoints - the UAC that originally targeted a SIPS URI and =
the
>> UAS or destination proxy advertising the URI - can collaborate to =
make
>> sure SIPS is requested and expected.
>>=20
>> In highly mediated networks where SIP requests transit many hops, I'm =
sure
>> SIPS is at best a nuisance.=20
>=20
> RS> That is an understatement.  SIPS is not used in virtually any =
component of the carrier based Public SIP Networks that rely on E.164 =
naming that I know of. Though it is specified as MUST support in the SIP =
Forum SIPConnect profiles for PBX to SP on the wire integration to my =
knowledge it has never been implemented by any vendor or SSP.  In short =
SIPS is a total Pain in the Ass.  Given the appalling track record the =
IETF has in designing implementable security protocols, I question =
whether this effort is worth the time.  But that=E2=80=99s OK ..if =
people want to tilt at windmills, be my guest.  What I want to know is =
does anyone really care?=20

My point is that a lot of people falsely believe that SIPS: is the way =
to go for SIP security and do not do=20
anything and certainly doesn=E2=80=99t even try to understand the =
complexity of SIPS:

We can start with removing SIPS: or start in the other end with SIP =
security. Media seems to be
the target of SIPBRANDY but my feeling is that SIP is an issue.

As stated many times, this was a problem we discovered hard at SIPit - =
where implementations registred
contacts using =E2=80=9C;transport=3DTLS=E2=80=9D and not using SIPS: - =
and did not have any client certificates.

It=E2=80=99s a mess and I feel that we need to clean this up in orders =
for developers to know where to go.

/O

>=20
>=20
> Its design optimized more for a trapezoid, or
>> something close to it, than a PSTN-like network. I'd hesitate though =
to
>> eliminate a signaling confidentiality mode for SIP that has some
>> possibility of preventing pervasive metadata collection, even if we =
know
>> there are some environments where that protection will not be =
provided.
>> HTTPS is a nuisance in many highly mediated parts of the web, too.
>>=20
>> Jon Peterson
>> Neustar, Inc.
>>=20
>> On 6/13/16, 6:56 AM, "sipcore on behalf of Drage, Keith (Nokia - GB)"
>> <sipcore-bounces@ietf.org on behalf of keith.drage@nokia.com> wrote:
>>=20
>>> That option was clearly understood by the SIP WG when we did RFC =
5630.
>>> However it was decided to very clearly state what it meant so that =
both
>>> service providers and users could decide whether to use it or not, =
and
>>> they still have that freedom. I suspect the majority have chosen not =
to,
>>> but the option is still there.
>>>=20
>>> Given that we spent so much time on providing RFC 5630 to update RFC =
3261
>>> in this area, I am reluctant to change the direction now - this =
achieved
>>> IETF consensus.
>>>=20
>>> Keith
>>>=20
>>> P.S. I also take this opportunity to remember Francois Audet, who =
did
>>> such a lot of work driving this document to completion.
>>>=20
>>> -----Original Message-----
>>> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of DOLLY,
>>> MARTIN C
>>> Sent: 12 June 2016 09:03
>>> To: Paul Kyzivat
>>> Cc: sipcore@ietf.org
>>> Subject: Re: [sipcore] SIPS must die, die, die
>>>=20
>>> I vote 3, don't use SIPS
>>>=20
>>> Martin C Dolly
>>> Lead Member of Technical Staff
>>> Core & Government/Regulatory Standards
>>> AT&T
>>> Cell: 609-903-3360
>>> Email: md3135@att.com
>>>=20
>>>> On Jun 11, 2016, at 1:00 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> =
wrote:
>>>>=20
>>>>> On 6/11/16 2:47 AM, Christer Holmberg wrote:
>>>>> Hi,
>>>>>=20
>>>>>>> I think it's time to spend some time with TLS in SIPcore again.
>>>>>>> What do you think?
>>>>>>=20
>>>>>> My memory is that the latest revision decided, as Keith says, =
that
>>>>>> if you make a call to a sips: URI, you > are guaranteed that =
every
>>>>>> signaling step is covered by TLS, or the call is rejected
>>>>>=20
>>>>> ...or the SIPS URI is "converted" to a SIP URI by an intermediary. =
In
>>>>> reality there is no guarantee.
>>>>=20
>>>> So the choices are:
>>>>=20
>>>> 1) use SIPS as defined, meeting all the constraints
>>>> 2) use SIPS but cheat when convenient
>>>> 3) don't use SIPS
>>>>=20
>>>> Can anyone cite are real deployments that meet (1)? I suspect there =
are
>>>> none.
>>>>=20
>>>> I hope nobody here is advocating (2). It has no useful security
>>>> properties.
>>>>=20
>>>> Hence ISTM that (3) is the only deployable alternative.
>>>>=20
>>>>   Thanks,
>>>>   Paul
>>>>=20
>>>> _______________________________________________
>>>> sipcore mailing list
>>>> sipcore@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>=20
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>=20
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>=20
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From nobody Tue Jun 14 01:49:03 2016
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Date: Tue, 14 Jun 2016 09:48:44 +0100
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Subject: Re: [sipcore] SIPS must die, die, die
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On 14 Jun 2016, at 8:43, Olle E. Johansson wrote:

> RS> That is an understatement.  SIPS is not used in virtually any 
> component of the carrier based Public SIP Networks that rely on E.164 
> naming that I know of. Though it is specified as MUST support in the 
> SIP Forum SIPConnect profiles for PBX to SP on the wire integration to 
> my knowledge it has never been implemented by any vendor or SSP.  In 
> short SIPS is a total Pain in the Ass.  Given the appalling track 
> record the IETF has in designing implementable security protocols, I 
> question whether this effort is worth the time.  But thatâ€™s OK ..if 
> people want to tilt at windmills, be my guest.  What I want to know is 
> does anyone really care?
>
>
> My point is that a lot of people falsely believe that SIPS: is the way 
> to go for SIP security and do not do
> anything and certainly doesnâ€™t even try to understand the complexity 
> of SIPS:
>
> We can start with removing SIPS: or start in the other end with SIP 
> security. Media seems to be
> the target of SIPBRANDY but my feeling is that SIP is an issue.

SIPBRANDY will target SIP-signaled media. SIP security certainly matters 
for that. Whether or not the group has anything to say about SIPS is up 
to the group to decide.  (But my guess from the work so far is that the 
group will focus more on using STIR/4474bis to protect fingerprints, 
etc.)

>
> As stated many times, this was a problem we discovered hard at SIPit - 
> where implementations registred
> contacts using â€œ;transport=TLSâ€ and not using SIPS: - and did not 
> have any client certificates.
>
> Itâ€™s a mess and I feel that we need to clean this up in orders for 
> developers to know where to go.


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Olle E. Johansson" <oej@edvina.net>, "Peterson, Jon" <jon.peterson@neustar.biz>
Thread-Topic: [sipcore] SIPS must die, die, die
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Subject: Re: [sipcore] SIPS must die, die, die
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Hi,

...

>I think we need to start with =B3what if we remove SIPS=B2 and go back and
>see what=B9s missing and how we can fix that. A lot of people point to SIP=
S
>as
>the solution - but no one is implementing or using it.
>
>What is the problem it solves? Is there another way? What=B9s the status
>for=20
>S/MIME.

Nobody does that either. And, it wouldn=B9t guarantee anything either, woul=
d
it?

>I don=B9t think we=B9ll find that the answer is SIPS: as a URI scheme, but=
 the
>SIP over TLS transport is still very much needed, discovered by NAPTR/SRV
>and supported by DANE or some other PKI structure.

You can already do SIP over TLS using the SIP: scheme - just establish a
TLS connection and send your message :)

Regards,

Christer


>> Jon Peterson
>> Neustar, Inc.
>>=20
>> On 6/13/16, 6:56 AM, "sipcore on behalf of Drage, Keith (Nokia - GB)"
>> <sipcore-bounces@ietf.org on behalf of keith.drage@nokia.com> wrote:
>>=20
>>> That option was clearly understood by the SIP WG when we did RFC 5630.
>>> However it was decided to very clearly state what it meant so that both
>>> service providers and users could decide whether to use it or not, and
>>> they still have that freedom. I suspect the majority have chosen not
>>>to,
>>> but the option is still there.
>>>=20
>>> Given that we spent so much time on providing RFC 5630 to update RFC
>>>3261
>>> in this area, I am reluctant to change the direction now - this
>>>achieved
>>> IETF consensus.
>>>=20
>>> Keith
>>>=20
>>> P.S. I also take this opportunity to remember Francois Audet, who did
>>> such a lot of work driving this document to completion.
>>>=20
>>> -----Original Message-----
>>> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of DOLLY,
>>> MARTIN C
>>> Sent: 12 June 2016 09:03
>>> To: Paul Kyzivat
>>> Cc: sipcore@ietf.org
>>> Subject: Re: [sipcore] SIPS must die, die, die
>>>=20
>>> I vote 3, don't use SIPS
>>>=20
>>> Martin C Dolly
>>> Lead Member of Technical Staff
>>> Core & Government/Regulatory Standards
>>> AT&T
>>> Cell: 609-903-3360
>>> Email: md3135@att.com
>>>=20
>>>> On Jun 11, 2016, at 1:00 PM, Paul Kyzivat <pkyzivat@alum.mit.edu>
>>>>wrote:
>>>>=20
>>>>> On 6/11/16 2:47 AM, Christer Holmberg wrote:
>>>>> Hi,
>>>>>=20
>>>>>>> I think it's time to spend some time with TLS in SIPcore again.
>>>>>>> What do you think?
>>>>>>=20
>>>>>> My memory is that the latest revision decided, as Keith says, that
>>>>>> if you make a call to a sips: URI, you > are guaranteed that every
>>>>>> signaling step is covered by TLS, or the call is rejected
>>>>>=20
>>>>> ...or the SIPS URI is "converted" to a SIP URI by an intermediary. In
>>>>> reality there is no guarantee.
>>>>=20
>>>> So the choices are:
>>>>=20
>>>> 1) use SIPS as defined, meeting all the constraints
>>>> 2) use SIPS but cheat when convenient
>>>> 3) don't use SIPS
>>>>=20
>>>> Can anyone cite are real deployments that meet (1)? I suspect there
>>>>are
>>>> none.
>>>>=20
>>>> I hope nobody here is advocating (2). It has no useful security
>>>> properties.
>>>>=20
>>>> Hence ISTM that (3) is the only deployable alternative.
>>>>=20
>>>>   Thanks,
>>>>   Paul
>>>>=20
>>>> _______________________________________________
>>>> sipcore mailing list
>>>> sipcore@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>=20
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>=20
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>=20
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>
>_______________________________________________
>sipcore mailing list
>sipcore@ietf.org
>https://www.ietf.org/mailman/listinfo/sipcore


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To: Ben Campbell <ben@nostrum.com>
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Subject: Re: [sipcore] SIPS must die, die, die
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> On 14 Jun 2016, at 10:48, Ben Campbell <ben@nostrum.com> wrote:
>=20
>=20
>=20
> On 14 Jun 2016, at 8:43, Olle E. Johansson wrote:
>=20
>> RS> That is an understatement.  SIPS is not used in virtually any =
component of the carrier based Public SIP Networks that rely on E.164 =
naming that I know of. Though it is specified as MUST support in the SIP =
Forum SIPConnect profiles for PBX to SP on the wire integration to my =
knowledge it has never been implemented by any vendor or SSP.  In short =
SIPS is a total Pain in the Ass.  Given the appalling track record the =
IETF has in designing implementable security protocols, I question =
whether this effort is worth the time.  But that=E2=80=99s OK ..if =
people want to tilt at windmills, be my guest.  What I want to know is =
does anyone really care?
>>=20
>>=20
>> My point is that a lot of people falsely believe that SIPS: is the =
way to go for SIP security and do not do
>> anything and certainly doesn=E2=80=99t even try to understand the =
complexity of SIPS:
>>=20
>> We can start with removing SIPS: or start in the other end with SIP =
security. Media seems to be
>> the target of SIPBRANDY but my feeling is that SIP is an issue.
>=20
> SIPBRANDY will target SIP-signaled media. SIP security certainly =
matters for that. Whether or not the group has anything to say about =
SIPS is up to the group to decide.  (But my guess from the work so far =
is that the group will focus more on using STIR/4474bis to protect =
fingerprints, etc.)
That was my guess too :-)

/O
>=20
>>=20
>> As stated many times, this was a problem we discovered hard at SIPit =
- where implementations registred
>> contacts using =E2=80=9C;transport=3DTLS=E2=80=9D and not using SIPS: =
- and did not have any client certificates.
>>=20
>> It=E2=80=99s a mess and I feel that we need to clean this up in =
orders for developers to know where to go.
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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Subject: Re: [sipcore] SIPS must die, die, die
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> On 14 Jun 2016, at 10:48, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>=20
> Hi,
>=20
> ...
>=20
>> I think we need to start with =C2=B3what if we remove SIPS=C2=B2 and =
go back and
>> see what=C2=B9s missing and how we can fix that. A lot of people =
point to SIPS
>> as
>> the solution - but no one is implementing or using it.
>>=20
>> What is the problem it solves? Is there another way? What=C2=B9s the =
status
>> for=20
>> S/MIME.
>=20
> Nobody does that either. And, it wouldn=C2=B9t guarantee anything =
either, would
> it?
Well, it would get end2end security - but no one has thought about =
b2bua=E2=80=99s
and all those middleboxes in that scenario.=20
Right, I have only seen a bit of code in resiprocate but no usage.
>=20
>> I don=C2=B9t think we=C2=B9ll find that the answer is SIPS: as a URI =
scheme, but the
>> SIP over TLS transport is still very much needed, discovered by =
NAPTR/SRV
>> and supported by DANE or some other PKI structure.
>=20
> You can already do SIP over TLS using the SIP: scheme - just establish =
a
> TLS connection and send your message :)
Right. But the problem is getting the inbound request. Only Outbound =
solves
that and no one wants outbound right?
So that puts us back to my other proposal about half-outbound.

/O

>=20
> Regards,
>=20
> Christer
>=20
>=20
>>> Jon Peterson
>>> Neustar, Inc.
>>>=20
>>> On 6/13/16, 6:56 AM, "sipcore on behalf of Drage, Keith (Nokia - =
GB)"
>>> <sipcore-bounces@ietf.org on behalf of keith.drage@nokia.com> wrote:
>>>=20
>>>> That option was clearly understood by the SIP WG when we did RFC =
5630.
>>>> However it was decided to very clearly state what it meant so that =
both
>>>> service providers and users could decide whether to use it or not, =
and
>>>> they still have that freedom. I suspect the majority have chosen =
not
>>>> to,
>>>> but the option is still there.
>>>>=20
>>>> Given that we spent so much time on providing RFC 5630 to update =
RFC
>>>> 3261
>>>> in this area, I am reluctant to change the direction now - this
>>>> achieved
>>>> IETF consensus.
>>>>=20
>>>> Keith
>>>>=20
>>>> P.S. I also take this opportunity to remember Francois Audet, who =
did
>>>> such a lot of work driving this document to completion.
>>>>=20
>>>> -----Original Message-----
>>>> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of DOLLY,
>>>> MARTIN C
>>>> Sent: 12 June 2016 09:03
>>>> To: Paul Kyzivat
>>>> Cc: sipcore@ietf.org
>>>> Subject: Re: [sipcore] SIPS must die, die, die
>>>>=20
>>>> I vote 3, don't use SIPS
>>>>=20
>>>> Martin C Dolly
>>>> Lead Member of Technical Staff
>>>> Core & Government/Regulatory Standards
>>>> AT&T
>>>> Cell: 609-903-3360
>>>> Email: md3135@att.com
>>>>=20
>>>>> On Jun 11, 2016, at 1:00 PM, Paul Kyzivat <pkyzivat@alum.mit.edu>
>>>>> wrote:
>>>>>=20
>>>>>> On 6/11/16 2:47 AM, Christer Holmberg wrote:
>>>>>> Hi,
>>>>>>=20
>>>>>>>> I think it's time to spend some time with TLS in SIPcore again.
>>>>>>>> What do you think?
>>>>>>>=20
>>>>>>> My memory is that the latest revision decided, as Keith says, =
that
>>>>>>> if you make a call to a sips: URI, you > are guaranteed that =
every
>>>>>>> signaling step is covered by TLS, or the call is rejected
>>>>>>=20
>>>>>> ...or the SIPS URI is "converted" to a SIP URI by an =
intermediary. In
>>>>>> reality there is no guarantee.
>>>>>=20
>>>>> So the choices are:
>>>>>=20
>>>>> 1) use SIPS as defined, meeting all the constraints
>>>>> 2) use SIPS but cheat when convenient
>>>>> 3) don't use SIPS
>>>>>=20
>>>>> Can anyone cite are real deployments that meet (1)? I suspect =
there
>>>>> are
>>>>> none.
>>>>>=20
>>>>> I hope nobody here is advocating (2). It has no useful security
>>>>> properties.
>>>>>=20
>>>>> Hence ISTM that (3) is the only deployable alternative.
>>>>>=20
>>>>>  Thanks,
>>>>>  Paul
>>>>>=20
>>>>> _______________________________________________
>>>>> sipcore mailing list
>>>>> sipcore@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>>=20
>>>> _______________________________________________
>>>> sipcore mailing list
>>>> sipcore@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>>=20
>>>> _______________________________________________
>>>> sipcore mailing list
>>>> sipcore@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>=20
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>=20
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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On Mon, Jun 13, 2016 at 3:36 PM, Peterson, Jon <jon.peterson@neustar.biz>
wrote:

>
> Just to play back the general problem statement, I think we'd agree that
> of your three numbered possible solutions, #1 is more or less the assumed
> baseline SIP behavior - cases where, for example, the server would send a
> Digest challenge in the backwards direction to authenticate the user agen=
t
> and then make any authorization decision based on its own ACLs or what ha=
ve
> you. Your solutions #2 and #3 cover the two cases I think are interesting=
:
> SSO-like ones where relying parties trust an identity assertion provided =
by
> an IdP of some sort (#2) and cases where the identity information is not
> included at all (or is at least optional) and instead the relying party
> trusts traits or attributes or what have you provided by the IdP. I've
> argued that SIP has always been able to do #1, that we at least already
> have some ways to do #2, and that we've already done requirements for #3
> (RFC4484) but to date have gone no farther, and that it would be new work
> if we were to specify protocol for #3.
>
> So yes, the third option is where we need to focus. But the point of my
> last mail was to say that what we need to narrow our discussion down to
> what the attributes are that we imagine applications in these use cases
> will require to process SIP requests, primarily SIP INVITE requests. You
> say some high-level things below like that your third-option token "might
> grant the user access to multiple services" on the conference service whi=
ch
> is your relying party. In terms of concrete authorization traits for this
> use case, towards the very end you suggest attributes of a conference lik=
e
> the number of participants or the media types permitted in the conference=
.
>
> Radhika sent a helpful summary of conference control protocols to the lis=
t
> which surveyed mechanisms like CCMP, BFCP, and MRB, protocols which
> actually let you manage attributes of a conference like the number of
> participants and media types allowed.
>

This conference use case was an example for a SIP-based conference server
(RFC4579), just to describe the idea around the delegation of AuthNZ.

If the server is a CCMP-based conference, then you would obviously use that
to control and manage the conference server.




> The interesting thing about protocols like CCMP is that they are not, you
> know, actually transported by SIP. CCMP is transported by HTTP. Why?
> Because the semantics of conference control doesn't actually map onto SIP
> INVITE transactions very well. They map quite aptly to HTTP, and in the
> HTTP context, you can certainly use OAuth if you like.
>


Help me understand this part.

Why can an HTTP request transport a token to a resource server to be able
to access a specific resource, while SIP request cannot transport a similar
token to a SIP server to access a service?



> You even will find some text about Role-Based Access Control in the XCON
> (RFC6503) security consideration. Although you can use SIP and SDP to set
> up a BFCP connection (RFC4583), the security of BFCP relies on mutual TLS
> between its own endpoints. It is unclear to me what carrying attribute
> information around in a SIP INVITE would accomplish with regards to
> authorizing CCMP or BFCP transactions.
>

SIP INVITE will carry attributes to authorize that specific SIP request;
there is nothing in proposal #3 that suggest that SIP will carry
authorization for CCMP or any other protocol.


>
> So again, the point of my last mail was that the problem of authorizing
> SIP INVITEs is actually different from the problem of authorizing generic
> application transactions. SIP INVITEs still have more or less the semanti=
cs
> of a call setup message: they are requesting that a session be initiated.
> Conference control is not managed with SIP transactions, and the security
> of conference control protocols does not devolve to authorizing SIP
> transactions.
>

Well, RFC4579 defines a SIP Conference Control mechanism that allows you to
initiate and control a conference server using only SIP requests; so I am
not sure I understand this statement, unless we have different definitions
for the term =E2=80=9Cconference control=E2=80=9D.



In any case, my point was not to discuss conference servers specifically,
but to discuss the general idea of delegating both AuthN and AuthZ in a SIP
network.



Regards,

 Rifaat





> I suspect we'll find that most of the other security attributes that
> "applications" need to do their jobs similarly don't map onto the semanti=
cs
> of a SIP INVITE - but, I'm still open to entertaining examples that might
> cast doubt on that suspicion.
>
> Jon Peterson
> Neustar, Inc.
>
> From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
> Date: Thursday, June 9, 2016 at 10:59 AM
> To: Jon Peterson <jon.peterson@neustar.biz>
> Cc: Christer Holmberg <christer.holmberg@ericsson.com>, "sipcore@ietf.org=
"
> <sipcore@ietf.org>
> Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
>
> Hi Jon,
>
> Sorry for the delayed reply.
>
> Let's take the video conference example, and discuss the possible ways fo=
r the
> conference server to provide service.
>
>
>          Trust Domain 1              :    Internet     :  Trust Domain 2
>                                      :                 :
>                       +--------+     :                 :
>                       | IdP    |     :                 :
>         +-------------| AuthZ  |     :                 :
>         |             |        |     :                 :
>         |             +--------+     :                 :
>         |                 |          :                 :
>         |                 |          :                 :
>     +--------+        +--------+     :                 :     +--------+
>     |        |        | SIP    |     :                 :     | SIP App|
>     |   UA   |--------| Proxy  |-----:-----------------:-----| Video  |
>     |        |        |        |     :                 :     | Conference
>     +--------+        +--------+     :                 :     +--------+
>                                      :                 :
>
> Possible solutions:
>
> 1. Conference Server acts as an AuthN and AuthZ server. Requests to
> the server must provide an identity and credentials, and services will
> be provided based on locally stored information associated with
> the identity provided in the request.
>
> 2. Conference Server acts as an AuthZ server, but delegates the AuthN
> part. Requests to the server must provide an identity token, and services
> will be provided based on locally stored information associated with
> the identity provided in the token.
>
> 3. Conference Server delegates both AuthN and AuthZ parts. Requests to
> the server must provide an access token, which might or might not
> include information about the user requesting the service, and services
> will be provided based on information provided in the access token or as
> a result of an introspection step using the access token.
>
>
> Now, let's look at the 3rd option a bit more, and assume that the
> server provides the following types of conference services:
> 1. audio conference
> 2. audio/video conference
> 3. web conference
>
> or some combination of the above.
>
> Let's also assume that the server allows control over the number
> of participants allowed in each one of these conference types or
> combinations.
>
>
> Here is one way that I envisioned we could use to address this use case:
>
> When an enterprise user in the 1st trust domain wants to register with
> the SIP Proxy, it will have to authenticate to the IdP and as a result
> either the UA or the Proxy (depends on the UA's capabilities) will be
> provided with an AuthN and AuthZ tokens. This process could involve an
> interim step to allow only the proxy to get access to the tokens, or the
> tokens could be sent to the UA which will then use them to communicate wi=
th
> the proxy.
>
> The AuthN token is the identity of the user, and the AuthZ token has
> all the services granted to the user by the IdP/AuthZ Service.
>
> When the enterprise user needs access to the conference service, the
> UA will send an INVITE to the Proxy, and will include the AuthZ token if =
it
> is in possession of the tokens; otherwise, the proxy will add the
> appropriate token to the outgoing INVITE sent to the conference server.
>
> The scope of the AuthZ token might grant the user access to
> multiple services; to avoid sending that AuthZ token to all servers
> providing services granted to the user, the proxy could obtain a differen=
t
> token with limited scope that is specific to the service being provided
> (conference services in this case); there is a standard Token Exchange
> mechanism for that.
>
> The AuthZ token will include information about the type of service
> allowed (audio, audio/video, etc), and the limit on the number of
> participants. The AuthZ token might or might not have any information abo=
ut
> the user requesting the service, which depends on the level of privacy
> needed.
>
> When the conference server receives this token in the SIP INVITE, all
> it has to do is validate that the token was signed by an IdP that it has
> a trust relationship with, and provide the service requested by the
> AuthZ token. The conference server does not need the AuthN token to be ab=
le
> to provide a service.
>
> Regards,
>  Rifaat
>
>
> On Wed, May 25, 2016 at 3:06 PM, Peterson, Jon <jon.peterson@neustar.biz>
> wrote:
>
>>
>> Also, if we decided to separate the AuthN and AuthZ, then we might need
>> to carry more than one token.
>>
>>
>> I could see an argument that there is a need to carry the identity of th=
e
>> user in one header, and a reference to a place where you can get attribu=
tes
>> about the originating user in another header, say. That's more or less w=
hat
>> we envisioned the last time we went down this path, which led to the
>> RFC4484 requirements, largely informed by SAML (which I seem to recall 3=
GPP
>> was interested in at the time).
>>
>> But it's no accident that the work on RFC4484 never went beyond
>> requirements. Today, I think that much of the useful work that could be
>> done on the authorization question is in identifying which SIP
>> authorization policies are likely to depend on attributes as opposed to
>> just the identity of the end user. While we can readily conceive of
>> applications adjunct to communications that would require attributes, th=
e
>> challenge has been justifying why you would invoke those applications wi=
th
>> SIP rather than, say, HTTP.  The fundamental semantics of a SIP INVITE h=
ave
>> remained pretty close to those of placing a telephone call.  The
>> authorization policies of a telephone call are really different than tha=
t
>> of a web service, say, both in terms of how and why they are invoked, wh=
at
>> kinds of parties are involved, what forms redirection or cross-domain
>> interaction can occur. With appropriate caveats for other methods (notab=
ly
>> MESSAGE, SUBSCRIBE/NOTIFY. INFO), SIP requests have authorization policy
>> requirements that look a lot like call treatment policies.
>>
>> That much said, I am kind of interested in call treatment policies, and
>> their interaction with identity information. But just saying that there
>> might be a video conference service in the cloud, say, doesn't shed a lo=
t
>> of light for me. If there's a video conference, I'd ordinarily think you
>> just call into it. If the user who opens the conference bridge needs to =
be
>> charged, that's identity information - which in its usual form will conv=
ey
>> the individual user and their domain, like user@domain. If you need more
>> complex semantics than that, it's probably because you're doing somethin=
g
>> more complicated than just calling in to the conference. But it's less
>> clear to me why you'd use SIP for such cases. Drilling down deeper into
>> these use cases would help me understand where we're going, anyway - the
>> use cases in RFC4484 Section 4, which largely involve billing (including
>> when dealing with various protocol gateways) and resource management (bo=
th
>> preemption and priority), ultimately didn't seem sufficient to motivate
>> protocol work.
>>
>> Another issues in the need, in some cases, to avoid sending the token to
>> the UA; in this case there is a need for an interim step to get the
>> token(s) to the SIP proxy.
>>
>>
>> If this information is carried in headers, then presumably it could be
>> added, consumed, subtracted, or whatever by any relevant intermediaries.
>>
>> Jon Peterson
>> Neustar, Inc.
>>
>
>

--001a11415d8669129205353d0cd6
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On Mon, Jun 13, 2016 at 3:36 PM, Peterson, Jon <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:jon.peterson@neustar.biz" target=3D"_blank">jon.peterson@neusta=
r.biz</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"m=
argin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;borde=
r-left-color:rgb(204,204,204);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>Just to play back the general problem statement, I think we&#39;d agre=
e that of your three numbered possible solutions, #1 is more or less the as=
sumed baseline SIP behavior - cases where, for example, the server would se=
nd a Digest challenge in the backwards
 direction to authenticate the user agent and then make any authorization d=
ecision based on its own ACLs or what have you. Your solutions #2 and #3 co=
ver the two cases I think are interesting: SSO-like ones where relying part=
ies trust an identity assertion
 provided by an IdP of some sort (#2) and cases where the identity informat=
ion is not included at all (or is at least optional) and instead the relyin=
g party trusts traits or attributes or what have you provided by the IdP. I=
&#39;ve argued that SIP has always been
 able to do #1, that we at least already have some ways to do #2, and that =
we&#39;ve already done requirements for #3 (RFC4484) but to date have gone =
no farther, and that it would be new work if we were to specify protocol fo=
r #3.</div>
<div><br>
</div>
<div>So yes, the third option is where we need to focus. But the point of m=
y last mail was to say that what we need to narrow our discussion down to w=
hat the attributes are that we imagine applications in these use cases will=
 require to process SIP requests,
 primarily SIP INVITE requests. You say some high-level things below like t=
hat your third-option token &quot;might grant the user access to multiple s=
ervices&quot; on the conference service which is your relying party. In ter=
ms of concrete authorization traits for this
 use case, towards the very end you suggest attributes of a conference like=
 the number of participants or the media types permitted in the conference.=
</div>
<div><br>
</div>
<div>Radhika sent a helpful summary of conference control protocols to the =
list which surveyed mechanisms like CCMP, BFCP, and MRB, protocols which ac=
tually let you manage attributes of a conference like the number of partici=
pants and media types allowed. </div></div></blockquote><div><br></div><div=
><p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:in=
itial;background-repeat:initial"><font face=3D"arial, helvetica, sans-serif=
">This
conference use case was an example for a SIP-based conference server (RFC45=
79),
just to describe the idea around the delegation of AuthNZ.</font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial"><font face=3D"arial, helvetica, sans-serif"=
>If
the server is a CCMP-based conference, then you would obviously use that to
control and manage the conference server.<br></font></p><p class=3D"MsoNorm=
al" style=3D"margin-bottom:0.0001pt;background-image:initial;background-rep=
eat:initial"><span style=3D"font-family:Arial,sans-serif"><br></span></p></=
div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px =
0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-col=
or:rgb(204,204,204);padding-left:1ex"><div style=3D"word-wrap:break-word;co=
lor:rgb(0,0,0);font-size:14px;font-family:Calibri,sans-serif"><div>The
 interesting thing about protocols like CCMP is that they are not, you know=
, actually transported by SIP. CCMP is transported by HTTP. Why? Because th=
e semantics of conference control doesn&#39;t actually map onto SIP INVITE =
transactions very well. They map quite
 aptly to HTTP, and in the HTTP context, you can certainly use OAuth if you=
 like.</div></div></blockquote><div>=C2=A0</div><div><p class=3D"MsoNormal"=
 style=3D"margin-bottom:0.0001pt;background-image:initial;background-repeat=
:initial"><font face=3D"arial, helvetica, sans-serif">Help me understand
this part.</font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial"><font face=3D"arial, helvetica, sans-serif"=
>Why can an HTTP
request transport a token to a resource server to be able to access a speci=
fic resource,
while SIP request cannot transport a similar token to a SIP server to acces=
s a
service?</font></p></div><div><br></div><div>=C2=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;bo=
rder-left-style:solid;border-left-color:rgb(204,204,204);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>You even will find some text about Role-Based =
Access Control in the XCON (RFC6503) security consideration. Although you c=
an use SIP and SDP to set up a BFCP connection (RFC4583),
 the security of BFCP relies on mutual TLS between its own endpoints. It is=
 unclear to me what carrying attribute information around in a SIP INVITE w=
ould accomplish with regards to authorizing CCMP or BFCP transactions.</div=
></div></blockquote><div><br></div><div><p class=3D"MsoNormal" style=3D"mar=
gin-bottom:0.0001pt;background-image:initial;background-repeat:initial"><fo=
nt face=3D"arial, helvetica, sans-serif">SIP INVITE will carry
attributes to authorize that specific SIP request; there is nothing in prop=
osal
#3 that suggest that SIP will carry authorization for CCMP or any other
protocol.</font><span style=3D"font-family:Arial,sans-serif;font-size:12pt"=
></span></p></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=
=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;=
border-left-color:rgb(204,204,204);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><br>
</div>
<div>So again, the point of my last mail was that the problem of authorizin=
g SIP INVITEs is actually different from the problem of authorizing generic=
 application transactions. SIP INVITEs still have more or less the semantic=
s of a call setup message: they
 are requesting that a session be initiated. Conference control is not mana=
ged with SIP transactions, and the security of conference control protocols=
 does not devolve to authorizing SIP transactions. </div></div></blockquote=
><div><br></div><div><p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt=
;background-image:initial;background-repeat:initial"><font face=3D"arial, h=
elvetica, sans-serif">Well, RFC4579 defines
a SIP Conference Control mechanism that allows you to initiate and control =
a
conference server using only SIP requests; so I am not sure I understand th=
is
statement, unless we have different definitions for the term =E2=80=9Cconfe=
rence
control=E2=80=9D.</font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial"><font face=3D"arial, helvetica, sans-serif"=
>=C2=A0</font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial"><font face=3D"arial, helvetica, sans-serif"=
>In any case, my point
was not to discuss conference servers specifically, but to discuss the gene=
ral idea of
delegating both AuthN and AuthZ in a SIP network.</font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial"><font face=3D"arial, helvetica, sans-serif"=
>=C2=A0</font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial"><font face=3D"arial, helvetica, sans-serif"=
>Regards,</font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial"><font face=3D"arial, helvetica, sans-serif"=
>=C2=A0Rifaat</font></p><p class=3D"MsoNormal" style=3D"margin-bottom:0.000=
1pt;background-image:initial;background-repeat:initial"><span style=3D"font=
-size:12pt;font-family:Arial,sans-serif"><br></span></p><p class=3D"MsoNorm=
al" style=3D"margin-bottom:0.0001pt;background-image:initial;background-rep=
eat:initial"><span style=3D"font-size:12pt;font-family:Arial,sans-serif"><b=
r></span></p></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=
=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;=
border-left-color:rgb(204,204,204);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>I suspect we&#39;ll find that most of the other security attributes
 that &quot;applications&quot; need to do their jobs similarly don&#39;t ma=
p onto the semantics of a SIP INVITE - but, I&#39;m still open to entertain=
ing examples that might cast doubt on that suspicion.=C2=A0</div>
<div><br>
</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, June 9, 2016 at 10:=
59 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>Christer Holmberg &lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmb=
erg@ericsson.com</a>&gt;, &quot;<a href=3D"mailto:sipcore@ietf.org" target=
=3D"_blank">sipcore@ietf.org</a>&quot; &lt;<a href=3D"mailto:sipcore@ietf.o=
rg" target=3D"_blank">sipcore@ietf.org</a>&gt;<span class=3D""><br>
<span style=3D"font-weight:bold">Subject: </span>Re: [sipcore] SIP AuthNZ P=
roblem Statement - v3<br>
</span></div><div><div class=3D"h5">
<div><br>
</div>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div>
<div>
<div dir=3D"ltr"><font face=3D"monospace,monospace">Hi Jon,</font>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Sorry for the delayed reply.</font>=
</div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div>
<div>
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><font face=3D"monospace,monospace">Let&#39;s take the video conference=
 example, and discuss the possible ways for=C2=A0</font><span style=3D"font=
-family:monospace,monospace">the conference server to provide service.</spa=
n></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0T=
rust Domain 1 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0: =C2=A0 =C2=
=A0Internet =C2=A0 =C2=A0 : =C2=A0Trust Domain 2</font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 :</font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 +--------+ =C2=A0 =C2=A0 : =C2=A0=
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :</font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | IdP =C2=A0 =C2=A0| =C2=A0 =C2=
=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :</font></div=
>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 +------=
-------| AuthZ =C2=A0| =C2=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 :</font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0 =C2=A0| =C2=
=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :</fon=
t></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 +--------+ =C2=A0 =C2=A0 : =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :</font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0=
 =C2=A0 =C2=A0: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :</=
font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0=
 =C2=A0 =C2=A0: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :</=
font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 +--------+ =C2=A0 =C2=
=A0 =C2=A0 =C2=A0+--------+ =C2=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 : =C2=A0 =C2=A0 +--------+</font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=
=A0 =C2=A0| =C2=A0 =C2=A0 =C2=A0 =C2=A0| SIP =C2=A0 =C2=A0| =C2=A0 =C2=A0 :=
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : =C2=A0 =C2=A0 | =
SIP App|</font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 | =C2=A0 UA =C2=A0 |-=
-------| Proxy =C2=A0|-----:-----------------:-----| Video =C2=A0|</font></=
div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=
=A0 =C2=A0| =C2=A0 =C2=A0 =C2=A0 =C2=A0| =C2=A0 =C2=A0 =C2=A0 =C2=A0| =C2=
=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : =C2=
=A0 =C2=A0 | Conference</font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 +--------+ =C2=A0 =C2=
=A0 =C2=A0 =C2=A0+--------+ =C2=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 : =C2=A0 =C2=A0 +--------+</font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 :</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Possible solutions:</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">1. Conference Server acts as an Aut=
hN and AuthZ server. Requests to the=C2=A0server must provide an identity a=
nd credentials, and services will be=C2=A0provided based on locally stored =
information associated with the=C2=A0identity provided
 in the request.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">2. Conference Server acts as an Aut=
hZ server, but delegates the AuthN part.=C2=A0Requests to the server must p=
rovide an identity token, and services will=C2=A0be provided based on local=
ly stored information associated with the=C2=A0identity
 provided in the token.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">3. Conference Server delegates both=
 AuthN and AuthZ parts. Requests to the=C2=A0server must provide an access =
token, which might or might not include=C2=A0information about the user req=
uesting the service, and services will=C2=A0be provided
 based on information provided in the access token or as a=C2=A0result of a=
n introspection step using the access token.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Now, let&#39;s look at the 3rd opti=
on a bit more, and assume that the server=C2=A0provides the following types=
 of conference services:</font></div>
<div><font face=3D"monospace,monospace">1. audio conference</font></div>
<div><font face=3D"monospace,monospace">2. audio/video conference</font></d=
iv>
<div><font face=3D"monospace,monospace">3. web conference</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">or some=C2=A0combination=C2=A0of th=
e above.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Let&#39;s also assume that the serv=
er allows control over the number of=C2=A0participants allowed in each one =
of these conference types or combinations.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Here is one way that I envisioned w=
e could use to address this use case:</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
</div>
</div>
</div>
<font face=3D"monospace,monospace">When an enterprise user in the 1st trust=
 domain wants to register with the=C2=A0SIP Proxy, it will have to authenti=
cate to the IdP and as a result either=C2=A0the UA or the Proxy (depends on=
 the UA&#39;s capabilities) will be provided=C2=A0with
 an AuthN and AuthZ tokens. This process could involve an interim step=C2=
=A0to allow only the proxy to get access to the tokens, or the tokens could=
 be=C2=A0sent to the UA which will then use them to communicate with the pr=
oxy.</font>
<div><font face=3D"monospace,monospace"><br>
The AuthN token is the identity of the user, and the AuthZ token has all=C2=
=A0the services granted to the user by the IdP/AuthZ Service.<br>
<br>
When the enterprise user needs access to the conference service, the UA=C2=
=A0will send an INVITE to the Proxy, and will include the AuthZ token if it=
 is=C2=A0in possession of the tokens; otherwise, the proxy will add the app=
ropriate=C2=A0token to the outgoing INVITE sent
 to the conference server.<br>
<br>
The scope of the AuthZ token might grant the user access to multiple=C2=A0s=
ervices; to avoid sending that AuthZ token to all servers providing=C2=A0se=
rvices granted to the user, the proxy could obtain a different token with=
=C2=A0limited scope that is specific to the service
 being provided (conference=C2=A0services in this case); there is a standar=
d Token Exchange mechanism for=C2=A0that.<br>
<br>
The AuthZ token will include information about the type of service allowed=
=C2=A0(audio, audio/video, etc), and the limit on the number of participant=
s.=C2=A0The AuthZ token might or might not have any information about the u=
ser=C2=A0requesting the service, which depends on
 the level of privacy needed.<br>
<br>
When the conference server receives this token in the SIP INVITE, all it=C2=
=A0has to do is validate that the token was signed by an IdP that it has a=
=C2=A0trust relationship with, and provide the service requested by the Aut=
hZ=C2=A0token. The conference server does not need
 the AuthN token to be able to=C2=A0provide a service.</font>
<div>
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Regards,</font></div>
<div><font face=3D"monospace,monospace">=C2=A0Rifaat</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
</div>
</div>
</div>
</div>
</div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Wed, May 25, 2016 at 3:06 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-style:solid;border-left-color:rgb(204,204,204);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">
<span><span>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
</div>
</div>
</div>
</span><span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>Also, if we decided to separate the AuthN and AuthZ, then we might nee=
d to carry more than one token.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span>
<div>I could see an argument that there is a need to carry the identity of =
the user in one header, and a reference to a place where you can get attrib=
utes about the originating user in another header, say. That&#39;s more or =
less what we envisioned the last time
 we went down this path, which led to the RFC4484 requirements, largely inf=
ormed by SAML (which I seem to recall 3GPP was interested in at the time).<=
/div>
<div><br>
</div>
<div>But it&#39;s no accident that the work on RFC4484 never went beyond re=
quirements. Today, I think that much of the useful work that could be done =
on the authorization question is in identifying which SIP authorization pol=
icies are likely to depend on attributes
 as opposed to just the identity of the end user. While we can readily conc=
eive of applications adjunct to communications that would require attribute=
s, the challenge has been justifying why you would invoke those application=
s with SIP rather than, say, HTTP.
 =C2=A0The fundamental semantics of a SIP INVITE have remained pretty close=
 to those of placing a telephone call.=C2=A0 The authorization policies of =
a telephone call are really different than that of a web service, say, both=
 in terms of how and why they are invoked,
 what kinds of parties are involved, what forms redirection or cross-domain=
 interaction can occur. With appropriate caveats for other methods (notably=
 MESSAGE, SUBSCRIBE/NOTIFY. INFO), SIP requests have authorization policy r=
equirements that look a lot like
 call treatment policies.</div>
<div><br>
</div>
<div>That much said, I am kind of interested in call treatment policies, an=
d their interaction with identity information. But just saying that there m=
ight be a video conference service in the cloud, say, doesn&#39;t shed a lo=
t of light for me. If there&#39;s a video
 conference, I&#39;d ordinarily think you just call into it. If the user wh=
o opens the conference bridge needs to be charged, that&#39;s identity info=
rmation - which in its usual form will convey the individual user and their=
 domain, like user@domain. If you need more
 complex semantics than that, it&#39;s probably because you&#39;re doing so=
mething more complicated than just calling in to the conference. But it&#39=
;s less clear to me why you&#39;d use SIP for such cases. Drilling down dee=
per into these use cases would help me understand
 where we&#39;re going, anyway - the use cases in RFC4484 Section 4, which =
largely involve billing (including when dealing with various protocol gatew=
ays) and resource management (both preemption and priority), ultimately did=
n&#39;t seem sufficient to motivate protocol
 work.</div>
<span>
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>Another issues in the need, in some cases, to avoid sending the token =
to the UA; in this case there is a need for an interim step to get the toke=
n(s) to the SIP proxy.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span>
<div>If this information is carried in headers, then presumably it could be=
 added, consumed, subtracted, or whatever by any relevant intermediaries.=
=C2=A0</div>
<span><font color=3D"#888888">
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
</font></span></div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</blockquote>
</div></div></span>
</div>

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

--001a11415d8669129205353d0cd6--


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From: "Hutton, Andrew" <andrew.hutton@unify.com>
To: Richard Shockey <richard@shockey.us>, "Peterson, Jon" <jon.peterson@neustar.biz>, "Drage, Keith (Nokia - GB)" <keith.drage@nokia.com>, "DOLLY, MARTIN C" <md3135@att.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [sipcore] SIPS must die, die, die
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Thread-Topic: [sipcore] SIPS must die, die, die
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Subject: Re: [sipcore] SIPS must die, die, die
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From: "Drage, Keith (Nokia - GB)" <keith.drage@nokia.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: [sipcore] SIPS must die, die, die
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Date: Tue, 14 Jun 2016 16:04:02 +0000
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Subject: Re: [sipcore] SIPS must die, die, die
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From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Date: Tue, 14 Jun 2016 12:09:19 -0400
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To: "Drage, Keith (Nokia - GB)" <keith.drage@nokia.com>
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Subject: Re: [sipcore] SIPS must die, die, die
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I am with Keith here.
There is no problem with the SIPS recommendations as defined in the RFC, so
I am not sure on what basis we would deprecate that.
If you do not like it, do not implement it.

Regards,
 Rifaat


On Tue, Jun 14, 2016 at 12:04 PM, Drage, Keith (Nokia - GB) <
keith.drage@nokia.com> wrote:

> Please confirm whether RFC 5630 (including annex A) is clear in this
> respect or not, and be specific about where it is not. You should not be
> implementing RFC 3261 without also implementing RFC 5630, as one updates
> the other.
>
> As far as I am concerned, that document is clear about what is needed to
> support SIPS, and what is needed to support transport=TLS.
>
> The use of SIPS should not generate a bug report, because it is not an
> error. You may need to reject the call because you cannot support the
> capability, but that is no different to many other reasons for rejection.
>
> We do not normally deprecate RFCs just because people have failed to read
> the document, of failed to provide their implementations in conformance
> with the document, or even just failed to implement a particular capability.
>
> If you don't want to support SIPS, then fine, that is an acceptable
> position, but you need more evidence to tell the world not to use it.
>
> Keith
>
> -----Original Message-----
> From: Hutton, Andrew [mailto:andrew.hutton@unify.com]
> Sent: 14 June 2016 16:51
> To: Richard Shockey; Peterson, Jon; Drage, Keith (Nokia - GB); DOLLY,
> MARTIN C; Paul Kyzivat
> Cc: sipcore@ietf.org
> Subject: RE: [sipcore] SIPS must die, die, die
>
> ON: 13 June 2016 Richard Shockey
>
> > Though it is specified as MUST support in the SIP Forum SIPConnect
> > profiles for PBX to SP on the wire integration to my knowledge it has
> > never been implemented by any vendor or SSP.  In short SIPS is a total
> > Pain in the Ass
>
> SIPS is actually not required by SIPconnect although TLS is a MUST
> requirement hence the confusion.
>
> Whenever I have seen SIPS used it has resulted in some kind of bug report
> being generated and I have to say I have not heard of it being used for
> some time.  It needs to deprecated and the use of TLS clarified.
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

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<div dir=3D"ltr">I am with Keith here.<div>There is no problem with the SIP=
S recommendations as defined in the RFC, so I am not sure on what basis we =
would deprecate that.=C2=A0</div><div>If you do not like it, do not impleme=
nt it.</div><div><br></div><div>Regards,</div><div>=C2=A0Rifaat</div><div><=
br></div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On=
 Tue, Jun 14, 2016 at 12:04 PM, Drage, Keith (Nokia - GB) <span dir=3D"ltr"=
>&lt;<a href=3D"mailto:keith.drage@nokia.com" target=3D"_blank">keith.drage=
@nokia.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">Please c=
onfirm whether RFC 5630 (including annex A) is clear in this respect or not=
, and be specific about where it is not. You should not be implementing RFC=
 3261 without also implementing RFC 5630, as one updates the other.<br>
<br>
As far as I am concerned, that document is clear about what is needed to su=
pport SIPS, and what is needed to support transport=3DTLS.<br>
<br>
The use of SIPS should not generate a bug report, because it is not an erro=
r. You may need to reject the call because you cannot support the capabilit=
y, but that is no different to many other reasons for rejection.<br>
<br>
We do not normally deprecate RFCs just because people have failed to read t=
he document, of failed to provide their implementations in conformance with=
 the document, or even just failed to implement a particular capability.<br=
>
<br>
If you don&#39;t want to support SIPS, then fine, that is an acceptable pos=
ition, but you need more evidence to tell the world not to use it.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Keith<br>
</font></span><span class=3D"im HOEnZb"><br>
-----Original Message-----<br>
From: Hutton, Andrew [mailto:<a href=3D"mailto:andrew.hutton@unify.com">and=
rew.hutton@unify.com</a>]<br>
Sent: 14 June 2016 16:51<br>
To: Richard Shockey; Peterson, Jon; Drage, Keith (Nokia - GB); DOLLY, MARTI=
N C; Paul Kyzivat<br>
Cc: <a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
</span><div class=3D"HOEnZb"><div class=3D"h5">Subject: RE: [sipcore] SIPS =
must die, die, die<br>
<br>
ON: 13 June 2016 Richard Shockey<br>
<br>
&gt; Though it is specified as MUST support in the SIP Forum SIPConnect<br>
&gt; profiles for PBX to SP on the wire integration to my knowledge it has<=
br>
&gt; never been implemented by any vendor or SSP.=C2=A0 In short SIPS is a =
total<br>
&gt; Pain in the Ass<br>
<br>
SIPS is actually not required by SIPconnect although TLS is a MUST requirem=
ent hence the confusion.<br>
<br>
Whenever I have seen SIPS used it has resulted in some kind of bug report b=
eing generated and I have to say I have not heard of it being used for some=
 time.=C2=A0 It needs to deprecated and the use of TLS clarified.<br>
<br>
_______________________________________________<br>
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<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" rel=3D"noreferrer=
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</div></div></blockquote></div><br></div>

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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] SIPS must die, die, die
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On 6/14/16 12:04 PM, Drage, Keith (Nokia - GB) wrote:
> Please confirm whether RFC 5630 (including annex A) is clear in this respect or not, and be specific about where it is not. You should not be implementing RFC 3261 without also implementing RFC 5630, as one updates the other.
>
> As far as I am concerned, that document is clear about what is needed to support SIPS, and what is needed to support transport=TLS.

I just looked at it.

An area that I would be concerned with is in topologies including SBCs. 
They aren't explicitly addressed, but that isn't surprising since almost 
none of the sip specs do. They do seem to be covered by section 5.1.1.3 
on Derived Dialogs. But I don't know if those implementing or deploying 
SBCs would agree on that. If somebody doesn't think this applies to 
SBCs, they they might be treated as a "get out of jail free" card to 
stop using TLS.

Another possible concern is calls that transition from SIPS to some 
non-SIP protocol. E.g., is it permissible to terminate a SIPS call on a 
PSTN gateway? I can't find anything in 5630 that considers these. On one 
had, I might really want the SIPS level security all the way to the 
gateway. On the other hand, if that works I might be deluded in thinking 
I have more security than is delivered by the PSTN.

> The use of SIPS should not generate a bug report, because it is not an error. You may need to reject the call because you cannot support the capability, but that is no different to many other reasons for rejection.
>
> We do not normally deprecate RFCs just because people have failed to read the document, of failed to provide their implementations in conformance with the document, or even just failed to implement a particular capability.
>
> If you don't want to support SIPS, then fine, that is an acceptable position, but you need more evidence to tell the world not to use it.

I have mixed feelings about this. ISTM the main problem is that you 
can't accurately claim that you support 3261 unless you build in code 
paths to support SIPS. But you are free to restrict your code to 
deployments that don't *use* SIPS. You might not ever test the SIPS 
paths (or even have a way to test them). What is the point of this?

And if you are building UAC that connects via registration you will 
never be able to use SIPS unless you also implement and use Outbound. 
But Outbound support is optional.

Perhaps the answer is to remove the mandatory to implement aspect of SIPS.

	Thanks,
	Paul

> Keith
>
> -----Original Message-----
> From: Hutton, Andrew [mailto:andrew.hutton@unify.com]
> Sent: 14 June 2016 16:51
> To: Richard Shockey; Peterson, Jon; Drage, Keith (Nokia - GB); DOLLY, MARTIN C; Paul Kyzivat
> Cc: sipcore@ietf.org
> Subject: RE: [sipcore] SIPS must die, die, die
>
> ON: 13 June 2016 Richard Shockey
>
>> Though it is specified as MUST support in the SIP Forum SIPConnect
>> profiles for PBX to SP on the wire integration to my knowledge it has
>> never been implemented by any vendor or SSP.  In short SIPS is a total
>> Pain in the Ass
>
> SIPS is actually not required by SIPconnect although TLS is a MUST requirement hence the confusion.
>
> Whenever I have seen SIPS used it has resulted in some kind of bug report being generated and I have to say I have not heard of it being used for some time.  It needs to deprecated and the use of TLS clarified.
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>


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To: "Drage, Keith (Nokia - GB)" <keith.drage@nokia.com>
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Cc: "Hutton, Andrew" <andrew.hutton@unify.com>, "sipcore@ietf.org" <sipcore@ietf.org>, Olle E Johansson <oej@edvina.net>
Subject: Re: [sipcore] SIPS must die, die, die
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> On 14 Jun 2016, at 18:04, Drage, Keith (Nokia - GB) =
<keith.drage@nokia.com> wrote:
>=20
> Please confirm whether RFC 5630 (including annex A) is clear in this =
respect or not, and be specific about where it is not. You should not be =
implementing RFC 3261 without also implementing RFC 5630, as one updates =
the other.
>=20
> As far as I am concerned, that document is clear about what is needed =
to support SIPS, and what is needed to support transport=3DTLS.
Yes, it points clearly to SIP Outbound (RFC 5626). Which also has very =
few implementations.
As an alternative it points to use of SIP client certs - which I can=E2=80=
=99t find any clear specification of.

>=20
> The use of SIPS should not generate a bug report, because it is not an =
error. You may need to reject the call because you cannot support the =
capability, but that is no different to many other reasons for =
rejection.
Agree.

>=20
> We do not normally deprecate RFCs just because people have failed to =
read the document, of failed to provide their implementations in =
conformance with the document, or even just failed to implement a =
particular capability.
>=20
> If you don't want to support SIPS, then fine, that is an acceptable =
position, but you need more evidence to tell the world not to use it.
I think the fact that it still there blocks usage of TLS. But let=E2=80=99=
s focus on proper use of TLS and how we can help that, because it seems
like most people agree we need to look into that again.

/O
>=20
> Keith
>=20
> -----Original Message-----
> From: Hutton, Andrew [mailto:andrew.hutton@unify.com]=20
> Sent: 14 June 2016 16:51
> To: Richard Shockey; Peterson, Jon; Drage, Keith (Nokia - GB); DOLLY, =
MARTIN C; Paul Kyzivat
> Cc: sipcore@ietf.org
> Subject: RE: [sipcore] SIPS must die, die, die
>=20
> ON: 13 June 2016 Richard Shockey=20
>=20
>> Though it is specified as MUST support in the SIP Forum SIPConnect=20
>> profiles for PBX to SP on the wire integration to my knowledge it has=20=

>> never been implemented by any vendor or SSP.  In short SIPS is a =
total=20
>> Pain in the Ass
>=20
> SIPS is actually not required by SIPconnect although TLS is a MUST =
requirement hence the confusion.
>=20
> Whenever I have seen SIPS used it has resulted in some kind of bug =
report being generated and I have to say I have not heard of it being =
used for some time.  It needs to deprecated and the use of TLS =
clarified.
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Cc: sipcore@ietf.org, Olle E Johansson <oej@edvina.net>
Subject: Re: [sipcore] SIPS must die, die, die
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> On 14 Jun 2016, at 18:37, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>=20
> On 6/14/16 12:04 PM, Drage, Keith (Nokia - GB) wrote:
>> Please confirm whether RFC 5630 (including annex A) is clear in this =
respect or not, and be specific about where it is not. You should not be =
implementing RFC 3261 without also implementing RFC 5630, as one updates =
the other.
>>=20
>> As far as I am concerned, that document is clear about what is needed =
to support SIPS, and what is needed to support transport=3DTLS.
>=20
> I just looked at it.
>=20
> An area that I would be concerned with is in topologies including =
SBCs. They aren't explicitly addressed, but that isn't surprising since =
almost none of the sip specs do. They do seem to be covered by section =
5.1.1.3 on Derived Dialogs. But I don't know if those implementing or =
deploying SBCs would agree on that. If somebody doesn't think this =
applies to SBCs, they they might be treated as a "get out of jail free" =
card to stop using TLS.
>=20
> Another possible concern is calls that transition from SIPS to some =
non-SIP protocol. E.g., is it permissible to terminate a SIPS call on a =
PSTN gateway? I can't find anything in 5630 that considers these. On one =
had, I might really want the SIPS level security all the way to the =
gateway. On the other hand, if that works I might be deluded in thinking =
I have more security than is delivered by the PSTN.
>=20
>> The use of SIPS should not generate a bug report, because it is not =
an error. You may need to reject the call because you cannot support the =
capability, but that is no different to many other reasons for =
rejection.
>>=20
>> We do not normally deprecate RFCs just because people have failed to =
read the document, of failed to provide their implementations in =
conformance with the document, or even just failed to implement a =
particular capability.
>>=20
>> If you don't want to support SIPS, then fine, that is an acceptable =
position, but you need more evidence to tell the world not to use it.
>=20
> I have mixed feelings about this. ISTM the main problem is that you =
can't accurately claim that you support 3261 unless you build in code =
paths to support SIPS. But you are free to restrict your code to =
deployments that don't *use* SIPS. You might not ever test the SIPS =
paths (or even have a way to test them). What is the point of this?
That=E2=80=99s a good point. If we remove SIPS: it=E2=80=99s no longer =
part of any requirements.
>=20
> And if you are building UAC that connects via registration you will =
never be able to use SIPS unless you also implement and use Outbound. =
But Outbound support is optional.
Well, the option is client certs) and mutual TLS auth, also discussed =
(and dismissed=E2=80=A6) in 5630  ;-)
>=20
> Perhaps the answer is to remove the mandatory to implement aspect of =
SIPS.
Yes.

/O
>=20
> 	Thanks,
> 	Paul
>=20
>> Keith
>>=20
>> -----Original Message-----
>> From: Hutton, Andrew [mailto:andrew.hutton@unify.com]
>> Sent: 14 June 2016 16:51
>> To: Richard Shockey; Peterson, Jon; Drage, Keith (Nokia - GB); DOLLY, =
MARTIN C; Paul Kyzivat
>> Cc: sipcore@ietf.org
>> Subject: RE: [sipcore] SIPS must die, die, die
>>=20
>> ON: 13 June 2016 Richard Shockey
>>=20
>>> Though it is specified as MUST support in the SIP Forum SIPConnect
>>> profiles for PBX to SP on the wire integration to my knowledge it =
has
>>> never been implemented by any vendor or SSP.  In short SIPS is a =
total
>>> Pain in the Ass
>>=20
>> SIPS is actually not required by SIPconnect although TLS is a MUST =
requirement hence the confusion.
>>=20
>> Whenever I have seen SIPS used it has resulted in some kind of bug =
report being generated and I have to say I have not heard of it being =
used for some time.  It needs to deprecated and the use of TLS =
clarified.
>>=20
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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From: "Rosen, Brian" <Brian.Rosen@neustar.biz>
To: Olle E Johansson <oej@edvina.net>
Thread-Topic: [sipcore] SIPS must die, die, die
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Subject: Re: [sipcore] SIPS must die, die, die
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Thread-Topic: [sipcore] SIP AuthNZ Problem Statement - v3
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Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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--_000_D3856F1719BE76jonpetersonneustarbiz_
Content-Type: text/plain; charset="Windows-1252"
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Radhika sent a helpful summary of conference control protocols to the list =
which surveyed mechanisms like CCMP, BFCP, and MRB, protocols which actuall=
y let you manage attributes of a conference like the number of participants=
 and media types allowed.

This conference use case was an example for a SIP-based conference server (=
RFC4579), just to describe the idea around the delegation of AuthNZ.
If the server is a CCMP-based conference, then you would obviously use that=
 to control and manage the conference server.

While RFC4579 (an implementation of the RFC4353 framework) provides means t=
o do conference control from within SIP, the kinds of control primitives it=
 specifies are totally decoupled from the sorts of policies that the attrib=
utes you proposed would constrain. Those policies fall under what RFC4353 c=
alls "conference policy." Neither RFC4579 nor its requirements (RFC4245) re=
ally say anything about how conference policy is set or distributed.


Authorizing SIP requests is misaligned with setting conference policy, espe=
cially in consideration of the sorts of attributes you seem to have in mind=
 - even when we're considering controlling conferences compliant with the R=
FC4353 framework.

That's because the conference in RFC4579 isn't "created" when a user agent =
sends an INVITE request to the conference bridge - it is created when the c=
onference URI is created, which is necessarily beforehand. Conferences in t=
he RF4353 framework are associated with conference policies,



The example of CCMP shows that you can do conference control with OAuth ove=
r in the HTTP realm without needing to add anything whatsoever to SIP.


The interesting thing about protocols like CCMP is that they are not, you k=
now, actually transported by SIP. CCMP is transported by HTTP. Why? Because=
 the semantics of conference control doesn't actually map onto SIP INVITE t=
ransactions very well. They map quite aptly to HTTP, and in the HTTP contex=
t, you can certainly use OAuth if you like.

Help me understand this part.
Why can an HTTP request transport a token to a resource server to be able t=
o access a specific resource, while SIP request cannot transport a similar =
token to a SIP server to access a service?

Of course both SIP and HTTP can carry arbitrary chunks of data, including a=
ny sort of token we can imagine. My point is that the semantics of an INVIT=
E are really different from the semantics of a POST - and that is why, ulti=
mately, protocols like CCMP don't use SIP as a transport but instead use HT=
TP.  INVITEs are sent by a user agent when it is trying to set up a media s=
ession between itself and some remote resource on the Internet. While RFC45=
79 has a way to set up a conference as a side effect of sending an INVITE t=
o connect to the conference, must of the subsequent control work happens to=
tally outside the context of that dialog, and the authorization of those op=
erations is completely disconnected from the authorization of the conferenc=
e-opening INVITE.

And of course, SIP can support faux-HTTP semantics with mechanisms like PUB=
LISH and NOTIFY. But I think the interesting question about standardizing a=
n approach to authorization of SIP requests is about INVITE.


You even will find some text about Role-Based Access Control in the XCON (R=
FC6503) security consideration. Although you can use SIP and SDP to set up =
a BFCP connection (RFC4583), the security of BFCP relies on mutual TLS betw=
een its own endpoints. It is unclear to me what carrying attribute informat=
ion around in a SIP INVITE would accomplish with regards to authorizing CCM=
P or BFCP transactions.

SIP INVITE will carry attributes to authorize that specific SIP request; th=
ere is nothing in proposal #3 that suggest that SIP will carry authorizatio=
n for CCMP or any other protocol.

That is where I think you and I have a disconnect. The SIP INVITE in the co=
nference control (RFC4579) case would not carry attributes like the ones yo=
u've identified in order to authorize that specific SIP request. The attrib=
utes that you identified, the media types the conference allows and the num=
ber of participants permitted simultaneously in a conference, have nothing =
to do with the SIP INVITE that opens the conference bridge. Those attribute=
s apply to authorization decisions regarding later participants that will b=
e added to the conference.

We can imagine that the



So again, the point of my last mail was that the problem of authorizing SIP=
 INVITEs is actually different from the problem of authorizing generic appl=
ication transactions. SIP INVITEs still have more or less the semantics of =
a call setup message: they are requesting that a session be initiated. Conf=
erence control is not managed with SIP transactions, and the security of co=
nference control protocols does not devolve to authorizing SIP transactions=
.

Well, RFC4579 defines a SIP Conference Control mechanism that allows you to=
 initiate and control a conference server using only SIP requests; so I am =
not sure I understand this statement, unless we have different definitions =
for the term =93conference control=94.

In any case, my point was not to discuss conference servers specifically, b=
ut to discuss the general idea of delegating both AuthN and AuthZ in a SIP =
network.

If you could furnish an example of attributes that would be carried in a SI=
P iNVITE that are actually salient to the authorization of that particular =
SIP request, we'd be getting a lot closer to common ground.

Jon Peterson
Neustar, Inc.


Regards,
 Rifaat



I suspect we'll find that most of the other security attributes that "appli=
cations" need to do their jobs similarly don't map onto the semantics of a =
SIP INVITE - but, I'm still open to entertaining examples that might cast d=
oubt on that suspicion.

Jon Peterson
Neustar, Inc.

From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com<mailto:rifaat.ietf@gmail.co=
m>>
Date: Thursday, June 9, 2016 at 10:59 AM
To: Jon Peterson <jon.peterson@neustar.biz<mailto:jon.peterson@neustar.biz>=
>
Cc: Christer Holmberg <christer.holmberg@ericsson.com<mailto:christer.holmb=
erg@ericsson.com>>, "sipcore@ietf.org<mailto:sipcore@ietf.org>" <sipcore@ie=
tf.org<mailto:sipcore@ietf.org>>
Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3

Hi Jon,

Sorry for the delayed reply.

Let's take the video conference example, and discuss the possible ways for =
the conference server to provide service.


         Trust Domain 1              :    Internet     :  Trust Domain 2
                                     :                 :
                      +--------+     :                 :
                      | IdP    |     :                 :
        +-------------| AuthZ  |     :                 :
        |             |        |     :                 :
        |             +--------+     :                 :
        |                 |          :                 :
        |                 |          :                 :
    +--------+        +--------+     :                 :     +--------+
    |        |        | SIP    |     :                 :     | SIP App|
    |   UA   |--------| Proxy  |-----:-----------------:-----| Video  |
    |        |        |        |     :                 :     | Conference
    +--------+        +--------+     :                 :     +--------+
                                     :                 :

Possible solutions:

1. Conference Server acts as an AuthN and AuthZ server. Requests to the ser=
ver must provide an identity and credentials, and services will be provided=
 based on locally stored information associated with the identity provided =
in the request.

2. Conference Server acts as an AuthZ server, but delegates the AuthN part.=
 Requests to the server must provide an identity token, and services will b=
e provided based on locally stored information associated with the identity=
 provided in the token.

3. Conference Server delegates both AuthN and AuthZ parts. Requests to the =
server must provide an access token, which might or might not include infor=
mation about the user requesting the service, and services will be provided=
 based on information provided in the access token or as a result of an int=
rospection step using the access token.


Now, let's look at the 3rd option a bit more, and assume that the server pr=
ovides the following types of conference services:
1. audio conference
2. audio/video conference
3. web conference

or some combination of the above.

Let's also assume that the server allows control over the number of partici=
pants allowed in each one of these conference types or combinations.


Here is one way that I envisioned we could use to address this use case:

When an enterprise user in the 1st trust domain wants to register with the =
SIP Proxy, it will have to authenticate to the IdP and as a result either t=
he UA or the Proxy (depends on the UA's capabilities) will be provided with=
 an AuthN and AuthZ tokens. This process could involve an interim step to a=
llow only the proxy to get access to the tokens, or the tokens could be sen=
t to the UA which will then use them to communicate with the proxy.

The AuthN token is the identity of the user, and the AuthZ token has all th=
e services granted to the user by the IdP/AuthZ Service.

When the enterprise user needs access to the conference service, the UA wil=
l send an INVITE to the Proxy, and will include the AuthZ token if it is in=
 possession of the tokens; otherwise, the proxy will add the appropriate to=
ken to the outgoing INVITE sent to the conference server.

The scope of the AuthZ token might grant the user access to multiple servic=
es; to avoid sending that AuthZ token to all servers providing services gra=
nted to the user, the proxy could obtain a different token with limited sco=
pe that is specific to the service being provided (conference services in t=
his case); there is a standard Token Exchange mechanism for that.

The AuthZ token will include information about the type of service allowed =
(audio, audio/video, etc), and the limit on the number of participants. The=
 AuthZ token might or might not have any information about the user request=
ing the service, which depends on the level of privacy needed.

When the conference server receives this token in the SIP INVITE, all it ha=
s to do is validate that the token was signed by an IdP that it has a trust=
 relationship with, and provide the service requested by the AuthZ token. T=
he conference server does not need the AuthN token to be able to provide a =
service.

Regards,
 Rifaat


On Wed, May 25, 2016 at 3:06 PM, Peterson, Jon <jon.peterson@neustar.biz<ma=
ilto:jon.peterson@neustar.biz>> wrote:

Also, if we decided to separate the AuthN and AuthZ, then we might need to =
carry more than one token.

I could see an argument that there is a need to carry the identity of the u=
ser in one header, and a reference to a place where you can get attributes =
about the originating user in another header, say. That's more or less what=
 we envisioned the last time we went down this path, which led to the RFC44=
84 requirements, largely informed by SAML (which I seem to recall 3GPP was =
interested in at the time).

But it's no accident that the work on RFC4484 never went beyond requirement=
s. Today, I think that much of the useful work that could be done on the au=
thorization question is in identifying which SIP authorization policies are=
 likely to depend on attributes as opposed to just the identity of the end =
user. While we can readily conceive of applications adjunct to communicatio=
ns that would require attributes, the challenge has been justifying why you=
 would invoke those applications with SIP rather than, say, HTTP.  The fund=
amental semantics of a SIP INVITE have remained pretty close to those of pl=
acing a telephone call.  The authorization policies of a telephone call are=
 really different than that of a web service, say, both in terms of how and=
 why they are invoked, what kinds of parties are involved, what forms redir=
ection or cross-domain interaction can occur. With appropriate caveats for =
other methods (notably MESSAGE, SUBSCRIBE/NOTIFY. INFO), SIP requests have =
authorization policy requirements that look a lot like call treatment polic=
ies.

That much said, I am kind of interested in call treatment policies, and the=
ir interaction with identity information. But just saying that there might =
be a video conference service in the cloud, say, doesn't shed a lot of ligh=
t for me. If there's a video conference, I'd ordinarily think you just call=
 into it. If the user who opens the conference bridge needs to be charged, =
that's identity information - which in its usual form will convey the indiv=
idual user and their domain, like user@domain. If you need more complex sem=
antics than that, it's probably because you're doing something more complic=
ated than just calling in to the conference. But it's less clear to me why =
you'd use SIP for such cases. Drilling down deeper into these use cases wou=
ld help me understand where we're going, anyway - the use cases in RFC4484 =
Section 4, which largely involve billing (including when dealing with vario=
us protocol gateways) and resource management (both preemption and priority=
), ultimately didn't seem sufficient to motivate protocol work.

Another issues in the need, in some cases, to avoid sending the token to th=
e UA; in this case there is a need for an interim step to get the token(s) =
to the SIP proxy.

If this information is carried in headers, then presumably it could be adde=
d, consumed, subtracted, or whatever by any relevant intermediaries.

Jon Peterson
Neustar, Inc.



--_000_D3856F1719BE76jonpetersonneustarbiz_
Content-Type: text/html; charset="Windows-1252"
Content-ID: <994E14D110FDD74780CA130274BD6076@neustar.biz>
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</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;">
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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>Radhika sent a helpful summary of conference control protocols to the =
list which surveyed mechanisms like CCMP, BFCP, and MRB, protocols which ac=
tually let you manage attributes of a conference like the number of partici=
pants and media types allowed.</div>
</div>
</blockquote>
<div><br>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">This conference use case was an e=
xample for a SIP-based conference server (RFC4579), just to describe the id=
ea around the delegation of AuthNZ.</font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">If the server is a CCMP-based con=
ference, then you would obviously use that to control and manage the confer=
ence server.<br>
</font></p>
</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>While RFC4579 (an implementation of the RFC4353 framework) provides me=
ans to do conference control from within SIP, the kinds of control primitiv=
es it specifies are totally decoupled from the sorts of policies that the a=
ttributes you proposed would constrain.
 Those policies fall under what RFC4353 calls &quot;conference policy.&quot=
; Neither RFC4579 nor its requirements (RFC4245) really say anything about =
how conference policy is set or distributed.</div>
<div><br>
</div>
<div><br>
</div>
<div>Authorizing SIP requests is misaligned with setting conference policy,=
 especially in consideration of the sorts of attributes you seem to have in=
 mind - even when we're considering controlling conferences compliant with =
the RFC4353 framework.&nbsp;</div>
<div><br>
</div>
<div>That's because the conference in RFC4579 isn't &quot;created&quot; whe=
n a user agent sends an INVITE request to the conference bridge - it is cre=
ated when the conference URI is created, which is necessarily beforehand. C=
onferences in the RF4353 framework are associated
 with conference policies,&nbsp;</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div>The example of CCMP shows that you can do conference control with OAut=
h over in the HTTP realm without needing to add anything whatsoever to SIP.=
&nbsp;</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<span style=3D"font-family:Arial,sans-serif"><br>
</span></p>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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>The interesting thing about protocols like CCMP is that they are not, =
you know, actually transported by SIP. CCMP is transported by HTTP. Why? Be=
cause the semantics of conference control doesn't actually map onto SIP INV=
ITE transactions very well. They
 map quite aptly to HTTP, and in the HTTP context, you can certainly use OA=
uth if you like.</div>
</div>
</blockquote>
<div>&nbsp;</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">Help me understand this part.</fo=
nt></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">Why can an HTTP request transport=
 a token to a resource server to be able to access a specific resource, whi=
le SIP request cannot transport a similar token to a SIP server to access a=
 service?</font></p>
</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>
<div>Of course both SIP and HTTP can carry arbitrary chunks of data, includ=
ing any sort of token we can imagine. My point is that the semantics of an =
INVITE are really different from the semantics of a POST - and that is why,=
 ultimately, protocols like CCMP
 don't use SIP as a transport but instead use HTTP. &nbsp;INVITEs are sent =
by a user agent when it is trying to set up a media session between itself =
and some remote resource on the Internet. While RFC4579 has a way to set up=
 a conference as a side effect of sending
 an INVITE to connect to the conference, must of the subsequent control wor=
k happens totally outside the context of that dialog, and the authorization=
 of those operations is completely disconnected from the authorization of t=
he conference-opening INVITE.</div>
<div><br>
</div>
<div>And of course, SIP can support faux-HTTP semantics with mechanisms lik=
e PUBLISH and NOTIFY. But I think the interesting question about standardiz=
ing an approach to authorization of SIP requests is about INVITE.</div>
</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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>You even will find some text about Role-Based Access Control in the XC=
ON (RFC6503) security consideration. Although you can use SIP and SDP to se=
t up a BFCP connection (RFC4583), the security of BFCP relies on mutual TLS=
 between its own endpoints. It is
 unclear to me what carrying attribute information around in a SIP INVITE w=
ould accomplish with regards to authorizing CCMP or BFCP transactions.</div=
>
</div>
</blockquote>
<div><br>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">SIP INVITE will carry attributes =
to authorize that specific SIP request; there is nothing in proposal #3 tha=
t suggest that SIP will carry authorization for CCMP or any other protocol.=
</font></p>
</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>That is where I think you and I have a disconnect. The SIP INVITE in t=
he conference control (RFC4579) case would not carry attributes like the on=
es you've identified in order to authorize that specific SIP request. The a=
ttributes that you identified, the
 media types the conference allows and the number of participants permitted=
 simultaneously in a conference, have nothing to do with the SIP INVITE tha=
t opens the conference bridge. Those attributes apply to authorization deci=
sions regarding later participants
 that will be added to the conference.</div>
<div><br>
</div>
<div>We can imagine that the&nbsp;</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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>So again, the point of my last mail was that the problem of authorizin=
g SIP INVITEs is actually different from the problem of authorizing generic=
 application transactions. SIP INVITEs still have more or less the semantic=
s of a call setup message: they
 are requesting that a session be initiated. Conference control is not mana=
ged with SIP transactions, and the security of conference control protocols=
 does not devolve to authorizing SIP transactions.
</div>
</div>
</blockquote>
<div><br>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">Well, RFC4579 defines a SIP Confe=
rence Control mechanism that allows you to initiate and control a conferenc=
e server using only SIP requests; so I am not sure I understand this statem=
ent, unless we have different definitions
 for the term =93conference control=94.</font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">&nbsp;</font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">In any case, my point was not to =
discuss conference servers specifically, but to discuss the general idea of=
 delegating both AuthN and AuthZ in a SIP network.</font></p>
</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>If you could furnish an example of attributes that would be carried in=
 a SIP iNVITE that are actually salient to the authorization of that partic=
ular SIP request, we'd be getting a lot closer to common ground.</div>
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">&nbsp;</font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">Regards,</font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">&nbsp;Rifaat</font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<span style=3D"font-size:12pt;font-family:Arial,sans-serif"><br>
</span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<span style=3D"font-size:12pt;font-family:Arial,sans-serif"><br>
</span></p>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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 suspect we'll find that most of the other security attributes that &=
quot;applications&quot; need to do their jobs similarly don't map onto the =
semantics of a SIP INVITE - but, I'm still open to entertaining examples th=
at might cast doubt on that suspicion.&nbsp;</div>
<div><br>
</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, June 9, 2016 at 10:=
59 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>Christer Holmberg &lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmb=
erg@ericsson.com</a>&gt;, &quot;<a href=3D"mailto:sipcore@ietf.org" target=
=3D"_blank">sipcore@ietf.org</a>&quot; &lt;<a href=3D"mailto:sipcore@ietf.o=
rg" target=3D"_blank">sipcore@ietf.org</a>&gt;<span class=3D""><br>
<span style=3D"font-weight:bold">Subject: </span>Re: [sipcore] SIP AuthNZ P=
roblem Statement - v3<br>
</span></div>
<div>
<div class=3D"h5">
<div><br>
</div>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div>
<div>
<div dir=3D"ltr"><font face=3D"monospace,monospace">Hi Jon,</font>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Sorry for the delayed reply.</font>=
</div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div>
<div>
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><font face=3D"monospace,monospace">Let's take the video conference exa=
mple, and discuss the possible ways for&nbsp;</font><span style=3D"font-fam=
ily:monospace,monospace">the conference server to provide service.</span></=
div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;T=
rust Domain 1 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;: &nbsp; &nbs=
p;Internet &nbsp; &nbsp; : &nbsp;Trust Domain 2</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &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; :</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &#43;--------&#43; &nbsp; &nbsp; =
: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | IdP &nbsp; &nbsp;| &nbsp; &nbsp=
; : &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; &#43;--=
-----------| AuthZ &nbsp;| &nbsp; &nbsp; : &nbsp; &nbsp; &nbsp; &nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; :</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; | &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; =
&nbsp; : &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</font></=
div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; | &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &#43;--------&#43; &nbsp; &nbsp; : &nb=
sp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; | &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &=
nbsp; &nbsp;: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</fo=
nt></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; | &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &=
nbsp; &nbsp;: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</fo=
nt></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &#43;--------&#43; &n=
bsp; &nbsp; &nbsp; &nbsp;&#43;--------&#43; &nbsp; &nbsp; : &nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : &nbsp; &nbsp; &#43;--------&#43;=
</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; | &nbsp; &nbsp; &nbsp=
; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp;| SIP &nbsp; &nbsp;| &nbsp; &nbsp; : &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : &nbsp; &nbsp; | SI=
P App|</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; | &nbsp; UA &nbsp; |-=
-------| Proxy &nbsp;|-----:-----------------:-----| Video &nbsp;|</font></=
div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; | &nbsp; &nbsp; &nbsp=
; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; =
&nbsp; : &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : &nbsp; &=
nbsp; | Conference</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &#43;--------&#43; &n=
bsp; &nbsp; &nbsp; &nbsp;&#43;--------&#43; &nbsp; &nbsp; : &nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : &nbsp; &nbsp; &#43;--------&#43;=
</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &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; :</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Possible solutions:</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">1. Conference Server acts as an Aut=
hN and AuthZ server. Requests to the&nbsp;server must provide an identity a=
nd credentials, and services will be&nbsp;provided based on locally stored =
information associated with the&nbsp;identity provided
 in the request.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">2. Conference Server acts as an Aut=
hZ server, but delegates the AuthN part.&nbsp;Requests to the server must p=
rovide an identity token, and services will&nbsp;be provided based on local=
ly stored information associated with the&nbsp;identity
 provided in the token.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">3. Conference Server delegates both=
 AuthN and AuthZ parts. Requests to the&nbsp;server must provide an access =
token, which might or might not include&nbsp;information about the user req=
uesting the service, and services will&nbsp;be provided
 based on information provided in the access token or as a&nbsp;result of a=
n introspection step using the access token.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Now, let's look at the 3rd option a=
 bit more, and assume that the server&nbsp;provides the following types of =
conference services:</font></div>
<div><font face=3D"monospace,monospace">1. audio conference</font></div>
<div><font face=3D"monospace,monospace">2. audio/video conference</font></d=
iv>
<div><font face=3D"monospace,monospace">3. web conference</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">or some&nbsp;combination&nbsp;of th=
e above.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Let's also assume that the server a=
llows control over the number of&nbsp;participants allowed in each one of t=
hese conference types or combinations.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Here is one way that I envisioned w=
e could use to address this use case:</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
</div>
</div>
</div>
<font face=3D"monospace,monospace">When an enterprise user in the 1st trust=
 domain wants to register with the&nbsp;SIP Proxy, it will have to authenti=
cate to the IdP and as a result either&nbsp;the UA or the Proxy (depends on=
 the UA's capabilities) will be provided&nbsp;with
 an AuthN and AuthZ tokens. This process could involve an interim step&nbsp=
;to allow only the proxy to get access to the tokens, or the tokens could b=
e&nbsp;sent to the UA which will then use them to communicate with the prox=
y.</font>
<div><font face=3D"monospace,monospace"><br>
The AuthN token is the identity of the user, and the AuthZ token has all&nb=
sp;the services granted to the user by the IdP/AuthZ Service.<br>
<br>
When the enterprise user needs access to the conference service, the UA&nbs=
p;will send an INVITE to the Proxy, and will include the AuthZ token if it =
is&nbsp;in possession of the tokens; otherwise, the proxy will add the appr=
opriate&nbsp;token to the outgoing INVITE sent
 to the conference server.<br>
<br>
The scope of the AuthZ token might grant the user access to multiple&nbsp;s=
ervices; to avoid sending that AuthZ token to all servers providing&nbsp;se=
rvices granted to the user, the proxy could obtain a different token with&n=
bsp;limited scope that is specific to the service
 being provided (conference&nbsp;services in this case); there is a standar=
d Token Exchange mechanism for&nbsp;that.<br>
<br>
The AuthZ token will include information about the type of service allowed&=
nbsp;(audio, audio/video, etc), and the limit on the number of participants=
.&nbsp;The AuthZ token might or might not have any information about the us=
er&nbsp;requesting the service, which depends on
 the level of privacy needed.<br>
<br>
When the conference server receives this token in the SIP INVITE, all it&nb=
sp;has to do is validate that the token was signed by an IdP that it has a&=
nbsp;trust relationship with, and provide the service requested by the Auth=
Z&nbsp;token. The conference server does not need
 the AuthN token to be able to&nbsp;provide a service.</font>
<div>
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Regards,</font></div>
<div><font face=3D"monospace,monospace">&nbsp;Rifaat</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
</div>
</div>
</div>
</div>
</div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Wed, May 25, 2016 at 3:06 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-style:solid;border-left-color:rgb(204,204,204);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">
<span><span>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
</div>
</div>
</div>
</span><span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>Also, if we decided to separate the AuthN and AuthZ, then we might nee=
d to carry more than one token.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span>
<div>I could see an argument that there is a need to carry the identity of =
the user in one header, and a reference to a place where you can get attrib=
utes about the originating user in another header, say. That's more or less=
 what we envisioned the last time
 we went down this path, which led to the RFC4484 requirements, largely inf=
ormed by SAML (which I seem to recall 3GPP was interested in at the time).<=
/div>
<div><br>
</div>
<div>But it's no accident that the work on RFC4484 never went beyond requir=
ements. Today, I think that much of the useful work that could be done on t=
he authorization question is in identifying which SIP authorization policie=
s are likely to depend on attributes
 as opposed to just the identity of the end user. While we can readily conc=
eive of applications adjunct to communications that would require attribute=
s, the challenge has been justifying why you would invoke those application=
s with SIP rather than, say, HTTP.
 &nbsp;The fundamental semantics of a SIP INVITE have remained pretty close=
 to those of placing a telephone call.&nbsp; The authorization policies of =
a telephone call are really different than that of a web service, say, both=
 in terms of how and why they are invoked,
 what kinds of parties are involved, what forms redirection or cross-domain=
 interaction can occur. With appropriate caveats for other methods (notably=
 MESSAGE, SUBSCRIBE/NOTIFY. INFO), SIP requests have authorization policy r=
equirements that look a lot like
 call treatment policies.</div>
<div><br>
</div>
<div>That much said, I am kind of interested in call treatment policies, an=
d their interaction with identity information. But just saying that there m=
ight be a video conference service in the cloud, say, doesn't shed a lot of=
 light for me. If there's a video
 conference, I'd ordinarily think you just call into it. If the user who op=
ens the conference bridge needs to be charged, that's identity information =
- which in its usual form will convey the individual user and their domain,=
 like user@domain. If you need more
 complex semantics than that, it's probably because you're doing something =
more complicated than just calling in to the conference. But it's less clea=
r to me why you'd use SIP for such cases. Drilling down deeper into these u=
se cases would help me understand
 where we're going, anyway - the use cases in RFC4484 Section 4, which larg=
ely involve billing (including when dealing with various protocol gateways)=
 and resource management (both preemption and priority), ultimately didn't =
seem sufficient to motivate protocol
 work.</div>
<span>
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>Another issues in the need, in some cases, to avoid sending the token =
to the UA; in this case there is a need for an interim step to get the toke=
n(s) to the SIP proxy.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span>
<div>If this information is carried in headers, then presumably it could be=
 added, consumed, subtracted, or whatever by any relevant intermediaries.&n=
bsp;</div>
<span><font color=3D"#888888">
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
</font></span></div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</blockquote>
</div>
</div>
</span></div>
</blockquote>
</div>
<br>
</div>
</div>
</blockquote>
</span>
</body>
</html>

--_000_D3856F1719BE76jonpetersonneustarbiz_--


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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Thread-Topic: [sipcore] SIP AuthNZ Problem Statement - v3
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Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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--_000_D38584BD19C0DFjonpetersonneustarbiz_
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Um, sorry, this mail got sent accidentally well before it was completed. Le=
t me try to repair it and resend.

Jon Peterson
Neustar, Inc.

From: Jon Peterson <jon.peterson@neustar.biz<mailto:jon.peterson@neustar.bi=
z>>
Date: Tuesday, June 14, 2016 at 9:52 AM
To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com<mailto:rifaat.ietf@gmail.com>=
>
Cc: Christer Holmberg <christer.holmberg@ericsson.com<mailto:christer.holmb=
erg@ericsson.com>>, "sipcore@ietf.org<mailto:sipcore@ietf.org>" <sipcore@ie=
tf.org<mailto:sipcore@ietf.org>>
Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3

Radhika sent a helpful summary of conference control protocols to the list =
which surveyed mechanisms like CCMP, BFCP, and MRB, protocols which actuall=
y let you manage attributes of a conference like the number of participants=
 and media types allowed.

This conference use case was an example for a SIP-based conference server (=
RFC4579), just to describe the idea around the delegation of AuthNZ.
If the server is a CCMP-based conference, then you would obviously use that=
 to control and manage the conference server.

While RFC4579 (an implementation of the RFC4353 framework) provides means t=
o do conference control from within SIP, the kinds of control primitives it=
 specifies are totally decoupled from the sorts of policies that the attrib=
utes you proposed would constrain. Those policies fall under what RFC4353 c=
alls "conference policy." Neither RFC4579 nor its requirements (RFC4245) re=
ally say anything about how conference policy is set or distributed.


Authorizing SIP requests is misaligned with setting conference policy, espe=
cially in consideration of the sorts of attributes you seem to have in mind=
 - even when we're considering controlling conferences compliant with the R=
FC4353 framework.

That's because the conference in RFC4579 isn't "created" when a user agent =
sends an INVITE request to the conference bridge - it is created when the c=
onference URI is created, which is necessarily beforehand. Conferences in t=
he RF4353 framework are associated with conference policies,



The example of CCMP shows that you can do conference control with OAuth ove=
r in the HTTP realm without needing to add anything whatsoever to SIP.


The interesting thing about protocols like CCMP is that they are not, you k=
now, actually transported by SIP. CCMP is transported by HTTP. Why? Because=
 the semantics of conference control doesn't actually map onto SIP INVITE t=
ransactions very well. They map quite aptly to HTTP, and in the HTTP contex=
t, you can certainly use OAuth if you like.

Help me understand this part.
Why can an HTTP request transport a token to a resource server to be able t=
o access a specific resource, while SIP request cannot transport a similar =
token to a SIP server to access a service?

Of course both SIP and HTTP can carry arbitrary chunks of data, including a=
ny sort of token we can imagine. My point is that the semantics of an INVIT=
E are really different from the semantics of a POST - and that is why, ulti=
mately, protocols like CCMP don't use SIP as a transport but instead use HT=
TP.  INVITEs are sent by a user agent when it is trying to set up a media s=
ession between itself and some remote resource on the Internet. While RFC45=
79 has a way to set up a conference as a side effect of sending an INVITE t=
o connect to the conference, must of the subsequent control work happens to=
tally outside the context of that dialog, and the authorization of those op=
erations is completely disconnected from the authorization of the conferenc=
e-opening INVITE.

And of course, SIP can support faux-HTTP semantics with mechanisms like PUB=
LISH and NOTIFY. But I think the interesting question about standardizing a=
n approach to authorization of SIP requests is about INVITE.


You even will find some text about Role-Based Access Control in the XCON (R=
FC6503) security consideration. Although you can use SIP and SDP to set up =
a BFCP connection (RFC4583), the security of BFCP relies on mutual TLS betw=
een its own endpoints. It is unclear to me what carrying attribute informat=
ion around in a SIP INVITE would accomplish with regards to authorizing CCM=
P or BFCP transactions.

SIP INVITE will carry attributes to authorize that specific SIP request; th=
ere is nothing in proposal #3 that suggest that SIP will carry authorizatio=
n for CCMP or any other protocol.

That is where I think you and I have a disconnect. The SIP INVITE in the co=
nference control (RFC4579) case would not carry attributes like the ones yo=
u've identified in order to authorize that specific SIP request. The attrib=
utes that you identified, the media types the conference allows and the num=
ber of participants permitted simultaneously in a conference, have nothing =
to do with the SIP INVITE that opens the conference bridge. Those attribute=
s apply to authorization decisions regarding later participants that will b=
e added to the conference.

We can imagine that the



So again, the point of my last mail was that the problem of authorizing SIP=
 INVITEs is actually different from the problem of authorizing generic appl=
ication transactions. SIP INVITEs still have more or less the semantics of =
a call setup message: they are requesting that a session be initiated. Conf=
erence control is not managed with SIP transactions, and the security of co=
nference control protocols does not devolve to authorizing SIP transactions=
.

Well, RFC4579 defines a SIP Conference Control mechanism that allows you to=
 initiate and control a conference server using only SIP requests; so I am =
not sure I understand this statement, unless we have different definitions =
for the term =93conference control=94.

In any case, my point was not to discuss conference servers specifically, b=
ut to discuss the general idea of delegating both AuthN and AuthZ in a SIP =
network.

If you could furnish an example of attributes that would be carried in a SI=
P iNVITE that are actually salient to the authorization of that particular =
SIP request, we'd be getting a lot closer to common ground.

Jon Peterson
Neustar, Inc.


Regards,
 Rifaat



I suspect we'll find that most of the other security attributes that "appli=
cations" need to do their jobs similarly don't map onto the semantics of a =
SIP INVITE - but, I'm still open to entertaining examples that might cast d=
oubt on that suspicion.

Jon Peterson
Neustar, Inc.

From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com<mailto:rifaat.ietf@gmail.co=
m>>
Date: Thursday, June 9, 2016 at 10:59 AM
To: Jon Peterson <jon.peterson@neustar.biz<mailto:jon.peterson@neustar.biz>=
>
Cc: Christer Holmberg <christer.holmberg@ericsson.com<mailto:christer.holmb=
erg@ericsson.com>>, "sipcore@ietf.org<mailto:sipcore@ietf.org>" <sipcore@ie=
tf.org<mailto:sipcore@ietf.org>>
Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3

Hi Jon,

Sorry for the delayed reply.

Let's take the video conference example, and discuss the possible ways for =
the conference server to provide service.


         Trust Domain 1              :    Internet     :  Trust Domain 2
                                     :                 :
                      +--------+     :                 :
                      | IdP    |     :                 :
        +-------------| AuthZ  |     :                 :
        |             |        |     :                 :
        |             +--------+     :                 :
        |                 |          :                 :
        |                 |          :                 :
    +--------+        +--------+     :                 :     +--------+
    |        |        | SIP    |     :                 :     | SIP App|
    |   UA   |--------| Proxy  |-----:-----------------:-----| Video  |
    |        |        |        |     :                 :     | Conference
    +--------+        +--------+     :                 :     +--------+
                                     :                 :

Possible solutions:

1. Conference Server acts as an AuthN and AuthZ server. Requests to the ser=
ver must provide an identity and credentials, and services will be provided=
 based on locally stored information associated with the identity provided =
in the request.

2. Conference Server acts as an AuthZ server, but delegates the AuthN part.=
 Requests to the server must provide an identity token, and services will b=
e provided based on locally stored information associated with the identity=
 provided in the token.

3. Conference Server delegates both AuthN and AuthZ parts. Requests to the =
server must provide an access token, which might or might not include infor=
mation about the user requesting the service, and services will be provided=
 based on information provided in the access token or as a result of an int=
rospection step using the access token.


Now, let's look at the 3rd option a bit more, and assume that the server pr=
ovides the following types of conference services:
1. audio conference
2. audio/video conference
3. web conference

or some combination of the above.

Let's also assume that the server allows control over the number of partici=
pants allowed in each one of these conference types or combinations.


Here is one way that I envisioned we could use to address this use case:

When an enterprise user in the 1st trust domain wants to register with the =
SIP Proxy, it will have to authenticate to the IdP and as a result either t=
he UA or the Proxy (depends on the UA's capabilities) will be provided with=
 an AuthN and AuthZ tokens. This process could involve an interim step to a=
llow only the proxy to get access to the tokens, or the tokens could be sen=
t to the UA which will then use them to communicate with the proxy.

The AuthN token is the identity of the user, and the AuthZ token has all th=
e services granted to the user by the IdP/AuthZ Service.

When the enterprise user needs access to the conference service, the UA wil=
l send an INVITE to the Proxy, and will include the AuthZ token if it is in=
 possession of the tokens; otherwise, the proxy will add the appropriate to=
ken to the outgoing INVITE sent to the conference server.

The scope of the AuthZ token might grant the user access to multiple servic=
es; to avoid sending that AuthZ token to all servers providing services gra=
nted to the user, the proxy could obtain a different token with limited sco=
pe that is specific to the service being provided (conference services in t=
his case); there is a standard Token Exchange mechanism for that.

The AuthZ token will include information about the type of service allowed =
(audio, audio/video, etc), and the limit on the number of participants. The=
 AuthZ token might or might not have any information about the user request=
ing the service, which depends on the level of privacy needed.

When the conference server receives this token in the SIP INVITE, all it ha=
s to do is validate that the token was signed by an IdP that it has a trust=
 relationship with, and provide the service requested by the AuthZ token. T=
he conference server does not need the AuthN token to be able to provide a =
service.

Regards,
 Rifaat


On Wed, May 25, 2016 at 3:06 PM, Peterson, Jon <jon.peterson@neustar.biz<ma=
ilto:jon.peterson@neustar.biz>> wrote:

Also, if we decided to separate the AuthN and AuthZ, then we might need to =
carry more than one token.

I could see an argument that there is a need to carry the identity of the u=
ser in one header, and a reference to a place where you can get attributes =
about the originating user in another header, say. That's more or less what=
 we envisioned the last time we went down this path, which led to the RFC44=
84 requirements, largely informed by SAML (which I seem to recall 3GPP was =
interested in at the time).

But it's no accident that the work on RFC4484 never went beyond requirement=
s. Today, I think that much of the useful work that could be done on the au=
thorization question is in identifying which SIP authorization policies are=
 likely to depend on attributes as opposed to just the identity of the end =
user. While we can readily conceive of applications adjunct to communicatio=
ns that would require attributes, the challenge has been justifying why you=
 would invoke those applications with SIP rather than, say, HTTP.  The fund=
amental semantics of a SIP INVITE have remained pretty close to those of pl=
acing a telephone call.  The authorization policies of a telephone call are=
 really different than that of a web service, say, both in terms of how and=
 why they are invoked, what kinds of parties are involved, what forms redir=
ection or cross-domain interaction can occur. With appropriate caveats for =
other methods (notably MESSAGE, SUBSCRIBE/NOTIFY. INFO), SIP requests have =
authorization policy requirements that look a lot like call treatment polic=
ies.

That much said, I am kind of interested in call treatment policies, and the=
ir interaction with identity information. But just saying that there might =
be a video conference service in the cloud, say, doesn't shed a lot of ligh=
t for me. If there's a video conference, I'd ordinarily think you just call=
 into it. If the user who opens the conference bridge needs to be charged, =
that's identity information - which in its usual form will convey the indiv=
idual user and their domain, like user@domain. If you need more complex sem=
antics than that, it's probably because you're doing something more complic=
ated than just calling in to the conference. But it's less clear to me why =
you'd use SIP for such cases. Drilling down deeper into these use cases wou=
ld help me understand where we're going, anyway - the use cases in RFC4484 =
Section 4, which largely involve billing (including when dealing with vario=
us protocol gateways) and resource management (both preemption and priority=
), ultimately didn't seem sufficient to motivate protocol work.

Another issues in the need, in some cases, to avoid sending the token to th=
e UA; in this case there is a need for an interim step to get the token(s) =
to the SIP proxy.

If this information is carried in headers, then presumably it could be adde=
d, consumed, subtracted, or whatever by any relevant intermediaries.

Jon Peterson
Neustar, Inc.



--_000_D38584BD19C0DFjonpetersonneustarbiz_
Content-Type: text/html; charset="Windows-1252"
Content-ID: <8D7F9C30B4770246AA28A0F67096FDC4@neustar.biz>
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</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>Um, sorry, this mail got sent accidentally well before it was complete=
d. Let me try to repair it and resend.</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>Jon Peterson &lt;<a href=3D"m=
ailto:jon.peterson@neustar.biz">jon.peterson@neustar.biz</a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Tuesday, June 14, 2016 at 9:5=
2 AM<br>
<span style=3D"font-weight:bold">To: </span>Rifaat Shekh-Yusef &lt;<a href=
=3D"mailto:rifaat.ietf@gmail.com">rifaat.ietf@gmail.com</a>&gt;<br>
<span style=3D"font-weight:bold">Cc: </span>Christer Holmberg &lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson.com</=
a>&gt;, &quot;<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a>&quot=
; &lt;<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [sipcore] SIP AuthNZ P=
roblem Statement - v3<br>
</div>
<div><br>
</div>
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div>
<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line=
-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-famil=
y: Calibri, sans-serif;">
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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>Radhika sent a helpful summary of conference control protocols to the =
list which surveyed mechanisms like CCMP, BFCP, and MRB, protocols which ac=
tually let you manage attributes of a conference like the number of partici=
pants and media types allowed.</div>
</div>
</blockquote>
<div><br>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">This conference use case was an e=
xample for a SIP-based conference server (RFC4579), just to describe the id=
ea around the delegation of AuthNZ.</font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">If the server is a CCMP-based con=
ference, then you would obviously use that to control and manage the confer=
ence server.<br>
</font></p>
</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>While RFC4579 (an implementation of the RFC4353 framework) provides me=
ans to do conference control from within SIP, the kinds of control primitiv=
es it specifies are totally decoupled from the sorts of policies that the a=
ttributes you proposed would constrain.
 Those policies fall under what RFC4353 calls &quot;conference policy.&quot=
; Neither RFC4579 nor its requirements (RFC4245) really say anything about =
how conference policy is set or distributed.</div>
<div><br>
</div>
<div><br>
</div>
<div>Authorizing SIP requests is misaligned with setting conference policy,=
 especially in consideration of the sorts of attributes you seem to have in=
 mind - even when we're considering controlling conferences compliant with =
the RFC4353 framework.&nbsp;</div>
<div><br>
</div>
<div>That's because the conference in RFC4579 isn't &quot;created&quot; whe=
n a user agent sends an INVITE request to the conference bridge - it is cre=
ated when the conference URI is created, which is necessarily beforehand. C=
onferences in the RF4353 framework are associated
 with conference policies,&nbsp;</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div>The example of CCMP shows that you can do conference control with OAut=
h over in the HTTP realm without needing to add anything whatsoever to SIP.=
&nbsp;</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<span style=3D"font-family:Arial,sans-serif"><br>
</span></p>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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>The interesting thing about protocols like CCMP is that they are not, =
you know, actually transported by SIP. CCMP is transported by HTTP. Why? Be=
cause the semantics of conference control doesn't actually map onto SIP INV=
ITE transactions very well. They
 map quite aptly to HTTP, and in the HTTP context, you can certainly use OA=
uth if you like.</div>
</div>
</blockquote>
<div>&nbsp;</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">Help me understand this part.</fo=
nt></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">Why can an HTTP request transport=
 a token to a resource server to be able to access a specific resource, whi=
le SIP request cannot transport a similar token to a SIP server to access a=
 service?</font></p>
</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>
<div>Of course both SIP and HTTP can carry arbitrary chunks of data, includ=
ing any sort of token we can imagine. My point is that the semantics of an =
INVITE are really different from the semantics of a POST - and that is why,=
 ultimately, protocols like CCMP
 don't use SIP as a transport but instead use HTTP. &nbsp;INVITEs are sent =
by a user agent when it is trying to set up a media session between itself =
and some remote resource on the Internet. While RFC4579 has a way to set up=
 a conference as a side effect of sending
 an INVITE to connect to the conference, must of the subsequent control wor=
k happens totally outside the context of that dialog, and the authorization=
 of those operations is completely disconnected from the authorization of t=
he conference-opening INVITE.</div>
<div><br>
</div>
<div>And of course, SIP can support faux-HTTP semantics with mechanisms lik=
e PUBLISH and NOTIFY. But I think the interesting question about standardiz=
ing an approach to authorization of SIP requests is about INVITE.</div>
</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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>You even will find some text about Role-Based Access Control in the XC=
ON (RFC6503) security consideration. Although you can use SIP and SDP to se=
t up a BFCP connection (RFC4583), the security of BFCP relies on mutual TLS=
 between its own endpoints. It is
 unclear to me what carrying attribute information around in a SIP INVITE w=
ould accomplish with regards to authorizing CCMP or BFCP transactions.</div=
>
</div>
</blockquote>
<div><br>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">SIP INVITE will carry attributes =
to authorize that specific SIP request; there is nothing in proposal #3 tha=
t suggest that SIP will carry authorization for CCMP or any other protocol.=
</font></p>
</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>That is where I think you and I have a disconnect. The SIP INVITE in t=
he conference control (RFC4579) case would not carry attributes like the on=
es you've identified in order to authorize that specific SIP request. The a=
ttributes that you identified, the
 media types the conference allows and the number of participants permitted=
 simultaneously in a conference, have nothing to do with the SIP INVITE tha=
t opens the conference bridge. Those attributes apply to authorization deci=
sions regarding later participants
 that will be added to the conference.</div>
<div><br>
</div>
<div>We can imagine that the&nbsp;</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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>So again, the point of my last mail was that the problem of authorizin=
g SIP INVITEs is actually different from the problem of authorizing generic=
 application transactions. SIP INVITEs still have more or less the semantic=
s of a call setup message: they
 are requesting that a session be initiated. Conference control is not mana=
ged with SIP transactions, and the security of conference control protocols=
 does not devolve to authorizing SIP transactions.
</div>
</div>
</blockquote>
<div><br>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">Well, RFC4579 defines a SIP Confe=
rence Control mechanism that allows you to initiate and control a conferenc=
e server using only SIP requests; so I am not sure I understand this statem=
ent, unless we have different definitions
 for the term =93conference control=94.</font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">&nbsp;</font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">In any case, my point was not to =
discuss conference servers specifically, but to discuss the general idea of=
 delegating both AuthN and AuthZ in a SIP network.</font></p>
</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>If you could furnish an example of attributes that would be carried in=
 a SIP iNVITE that are actually salient to the authorization of that partic=
ular SIP request, we'd be getting a lot closer to common ground.</div>
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">&nbsp;</font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">Regards,</font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">&nbsp;Rifaat</font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<span style=3D"font-size:12pt;font-family:Arial,sans-serif"><br>
</span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<span style=3D"font-size:12pt;font-family:Arial,sans-serif"><br>
</span></p>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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 suspect we'll find that most of the other security attributes that &=
quot;applications&quot; need to do their jobs similarly don't map onto the =
semantics of a SIP INVITE - but, I'm still open to entertaining examples th=
at might cast doubt on that suspicion.&nbsp;</div>
<div><br>
</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, June 9, 2016 at 10:=
59 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>Christer Holmberg &lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmb=
erg@ericsson.com</a>&gt;, &quot;<a href=3D"mailto:sipcore@ietf.org" target=
=3D"_blank">sipcore@ietf.org</a>&quot; &lt;<a href=3D"mailto:sipcore@ietf.o=
rg" target=3D"_blank">sipcore@ietf.org</a>&gt;<span class=3D""><br>
<span style=3D"font-weight:bold">Subject: </span>Re: [sipcore] SIP AuthNZ P=
roblem Statement - v3<br>
</span></div>
<div>
<div class=3D"h5">
<div><br>
</div>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div>
<div>
<div dir=3D"ltr"><font face=3D"monospace,monospace">Hi Jon,</font>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Sorry for the delayed reply.</font>=
</div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div>
<div>
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><font face=3D"monospace,monospace">Let's take the video conference exa=
mple, and discuss the possible ways for&nbsp;</font><span style=3D"font-fam=
ily:monospace,monospace">the conference server to provide service.</span></=
div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;T=
rust Domain 1 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;: &nbsp; &nbs=
p;Internet &nbsp; &nbsp; : &nbsp;Trust Domain 2</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &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; :</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &#43;--------&#43; &nbsp; &nbsp; =
: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | IdP &nbsp; &nbsp;| &nbsp; &nbsp=
; : &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; &#43;--=
-----------| AuthZ &nbsp;| &nbsp; &nbsp; : &nbsp; &nbsp; &nbsp; &nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; :</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; | &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; =
&nbsp; : &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</font></=
div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; | &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &#43;--------&#43; &nbsp; &nbsp; : &nb=
sp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; | &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &=
nbsp; &nbsp;: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</fo=
nt></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; | &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &=
nbsp; &nbsp;: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</fo=
nt></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &#43;--------&#43; &n=
bsp; &nbsp; &nbsp; &nbsp;&#43;--------&#43; &nbsp; &nbsp; : &nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : &nbsp; &nbsp; &#43;--------&#43;=
</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; | &nbsp; &nbsp; &nbsp=
; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp;| SIP &nbsp; &nbsp;| &nbsp; &nbsp; : &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : &nbsp; &nbsp; | SI=
P App|</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; | &nbsp; UA &nbsp; |-=
-------| Proxy &nbsp;|-----:-----------------:-----| Video &nbsp;|</font></=
div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; | &nbsp; &nbsp; &nbsp=
; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; =
&nbsp; : &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : &nbsp; &=
nbsp; | Conference</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &#43;--------&#43; &n=
bsp; &nbsp; &nbsp; &nbsp;&#43;--------&#43; &nbsp; &nbsp; : &nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : &nbsp; &nbsp; &#43;--------&#43;=
</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &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; :</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Possible solutions:</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">1. Conference Server acts as an Aut=
hN and AuthZ server. Requests to the&nbsp;server must provide an identity a=
nd credentials, and services will be&nbsp;provided based on locally stored =
information associated with the&nbsp;identity provided
 in the request.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">2. Conference Server acts as an Aut=
hZ server, but delegates the AuthN part.&nbsp;Requests to the server must p=
rovide an identity token, and services will&nbsp;be provided based on local=
ly stored information associated with the&nbsp;identity
 provided in the token.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">3. Conference Server delegates both=
 AuthN and AuthZ parts. Requests to the&nbsp;server must provide an access =
token, which might or might not include&nbsp;information about the user req=
uesting the service, and services will&nbsp;be provided
 based on information provided in the access token or as a&nbsp;result of a=
n introspection step using the access token.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Now, let's look at the 3rd option a=
 bit more, and assume that the server&nbsp;provides the following types of =
conference services:</font></div>
<div><font face=3D"monospace,monospace">1. audio conference</font></div>
<div><font face=3D"monospace,monospace">2. audio/video conference</font></d=
iv>
<div><font face=3D"monospace,monospace">3. web conference</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">or some&nbsp;combination&nbsp;of th=
e above.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Let's also assume that the server a=
llows control over the number of&nbsp;participants allowed in each one of t=
hese conference types or combinations.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Here is one way that I envisioned w=
e could use to address this use case:</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
</div>
</div>
</div>
<font face=3D"monospace,monospace">When an enterprise user in the 1st trust=
 domain wants to register with the&nbsp;SIP Proxy, it will have to authenti=
cate to the IdP and as a result either&nbsp;the UA or the Proxy (depends on=
 the UA's capabilities) will be provided&nbsp;with
 an AuthN and AuthZ tokens. This process could involve an interim step&nbsp=
;to allow only the proxy to get access to the tokens, or the tokens could b=
e&nbsp;sent to the UA which will then use them to communicate with the prox=
y.</font>
<div><font face=3D"monospace,monospace"><br>
The AuthN token is the identity of the user, and the AuthZ token has all&nb=
sp;the services granted to the user by the IdP/AuthZ Service.<br>
<br>
When the enterprise user needs access to the conference service, the UA&nbs=
p;will send an INVITE to the Proxy, and will include the AuthZ token if it =
is&nbsp;in possession of the tokens; otherwise, the proxy will add the appr=
opriate&nbsp;token to the outgoing INVITE sent
 to the conference server.<br>
<br>
The scope of the AuthZ token might grant the user access to multiple&nbsp;s=
ervices; to avoid sending that AuthZ token to all servers providing&nbsp;se=
rvices granted to the user, the proxy could obtain a different token with&n=
bsp;limited scope that is specific to the service
 being provided (conference&nbsp;services in this case); there is a standar=
d Token Exchange mechanism for&nbsp;that.<br>
<br>
The AuthZ token will include information about the type of service allowed&=
nbsp;(audio, audio/video, etc), and the limit on the number of participants=
.&nbsp;The AuthZ token might or might not have any information about the us=
er&nbsp;requesting the service, which depends on
 the level of privacy needed.<br>
<br>
When the conference server receives this token in the SIP INVITE, all it&nb=
sp;has to do is validate that the token was signed by an IdP that it has a&=
nbsp;trust relationship with, and provide the service requested by the Auth=
Z&nbsp;token. The conference server does not need
 the AuthN token to be able to&nbsp;provide a service.</font>
<div>
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Regards,</font></div>
<div><font face=3D"monospace,monospace">&nbsp;Rifaat</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
</div>
</div>
</div>
</div>
</div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Wed, May 25, 2016 at 3:06 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-style:solid;border-left-color:rgb(204,204,204);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">
<span><span>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
</div>
</div>
</div>
</span><span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>Also, if we decided to separate the AuthN and AuthZ, then we might nee=
d to carry more than one token.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span>
<div>I could see an argument that there is a need to carry the identity of =
the user in one header, and a reference to a place where you can get attrib=
utes about the originating user in another header, say. That's more or less=
 what we envisioned the last time
 we went down this path, which led to the RFC4484 requirements, largely inf=
ormed by SAML (which I seem to recall 3GPP was interested in at the time).<=
/div>
<div><br>
</div>
<div>But it's no accident that the work on RFC4484 never went beyond requir=
ements. Today, I think that much of the useful work that could be done on t=
he authorization question is in identifying which SIP authorization policie=
s are likely to depend on attributes
 as opposed to just the identity of the end user. While we can readily conc=
eive of applications adjunct to communications that would require attribute=
s, the challenge has been justifying why you would invoke those application=
s with SIP rather than, say, HTTP.
 &nbsp;The fundamental semantics of a SIP INVITE have remained pretty close=
 to those of placing a telephone call.&nbsp; The authorization policies of =
a telephone call are really different than that of a web service, say, both=
 in terms of how and why they are invoked,
 what kinds of parties are involved, what forms redirection or cross-domain=
 interaction can occur. With appropriate caveats for other methods (notably=
 MESSAGE, SUBSCRIBE/NOTIFY. INFO), SIP requests have authorization policy r=
equirements that look a lot like
 call treatment policies.</div>
<div><br>
</div>
<div>That much said, I am kind of interested in call treatment policies, an=
d their interaction with identity information. But just saying that there m=
ight be a video conference service in the cloud, say, doesn't shed a lot of=
 light for me. If there's a video
 conference, I'd ordinarily think you just call into it. If the user who op=
ens the conference bridge needs to be charged, that's identity information =
- which in its usual form will convey the individual user and their domain,=
 like user@domain. If you need more
 complex semantics than that, it's probably because you're doing something =
more complicated than just calling in to the conference. But it's less clea=
r to me why you'd use SIP for such cases. Drilling down deeper into these u=
se cases would help me understand
 where we're going, anyway - the use cases in RFC4484 Section 4, which larg=
ely involve billing (including when dealing with various protocol gateways)=
 and resource management (both preemption and priority), ultimately didn't =
seem sufficient to motivate protocol
 work.</div>
<span>
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>Another issues in the need, in some cases, to avoid sending the token =
to the UA; in this case there is a need for an interim step to get the toke=
n(s) to the SIP proxy.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span>
<div>If this information is carried in headers, then presumably it could be=
 added, consumed, subtracted, or whatever by any relevant intermediaries.&n=
bsp;</div>
<span><font color=3D"#888888">
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
</font></span></div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</blockquote>
</div>
</div>
</span></div>
</blockquote>
</div>
<br>
</div>
</div>
</blockquote>
</span></div>
</div>
</blockquote>
</span>
</body>
</html>

--_000_D38584BD19C0DFjonpetersonneustarbiz_--


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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Thread-Topic: [sipcore] SIP AuthNZ Problem Statement - v3
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Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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This time my response should be more complete. Sorry about that!


Radhika sent a helpful summary of conference control protocols to the list =
which surveyed mechanisms like CCMP, BFCP, and MRB, protocols which actuall=
y let you manage attributes of a conference like the number of participants=
 and media types allowed.

This conference use case was an example for a SIP-based conference server (=
RFC4579), just to describe the idea around the delegation of AuthNZ.
If the server is a CCMP-based conference, then you would obviously use that=
 to control and manage the conference server.

While RFC4579 (an implementation of the RFC4353 framework) provides a means=
 to do conference control from within SIP, the kinds of control primitives =
it specifies are totally decoupled from the sorts of policies that the attr=
ibutes you proposed would constrain. Those policies fall under what RFC4353=
 calls "conference policy." Neither RFC4579 nor its requirements (RFC4245) =
really say anything about how conference policy is set or distributed. RFC4=
353 does, though, and what it says is (in 4.6) is "the conference policy ca=
n be manipulated using web applications or voice applications [i.e. IVRs - =
JFP]. It can also be manipulated with non-SIP-specific standard or propriet=
ary protocols."

Authorizing SIP requests is misaligned with setting conference policy, espe=
cially in consideration of the sorts of attributes you seem to have in mind=
 - even when we're considering controlling conferences compliant with the R=
FC4353 framework.  That's because the conference in RFC4579 isn't "created"=
 when a user agent sends an INVITE request to the conference bridge - it is=
 created when the conference URI is created, which is necessarily beforehan=
d (or at best, ad hoc through the conference factory). Conference policy is=
 set and disseminated in that context, not in the context of the SIP transa=
ction that happens to open the bridge.

Moreover, my point was that the example of CCMP shows that you can do confe=
rence control with OAuth over in the HTTP realm without needing to add anyt=
hing whatsoever to SIP.  If you need OAuth-style controls for conferencing,=
 that would seem to be a quicker path to implementation.

The interesting thing about protocols like CCMP is that they are not, you k=
now, actually transported by SIP. CCMP is transported by HTTP. Why? Because=
 the semantics of conference control doesn't actually map onto SIP INVITE t=
ransactions very well. They map quite aptly to HTTP, and in the HTTP contex=
t, you can certainly use OAuth if you like.

Help me understand this part.
Why can an HTTP request transport a token to a resource server to be able t=
o access a specific resource, while SIP request cannot transport a similar =
token to a SIP server to access a service?

Of course both SIP and HTTP can carry arbitrary chunks of data, including a=
ny sort of token we can imagine. My point is that the semantics of an INVIT=
E are really different from the semantics of a POST - and that is why, ulti=
mately, protocols like CCMP don't use SIP as a transport but instead use HT=
TP.  INVITEs are sent by a user agent when it is trying to set up a media s=
ession between itself and some remote resource on the Internet. While RFC45=
79 has a way to open a conference as a side effect of sending an INVITE to =
connect to the conference, the policies associated with the sorts of attrib=
utes you specified are set totally outside the context of that dialog, and =
the execution of those policies is completely disconnected from the authori=
zation of the conference-opening INVITE. More on this below.

 I think the interesting question about standardizing an approach to author=
ization of SIP requests is about INVITE. Of course, SIP supports faux-HTTP =
semantics with mechanisms like PUBLISH and NOTIFY, and we could talk about =
authorizing a number of other non-dialog-forming methods as well, but if we=
're going to kick off an effort to specify authorization policy for SIP, th=
e primary thing we need to worry about is INVITE.


You even will find some text about Role-Based Access Control in the XCON (R=
FC6503) security consideration. Although you can use SIP and SDP to set up =
a BFCP connection (RFC4583), the security of BFCP relies on mutual TLS betw=
een its own endpoints. It is unclear to me what carrying attribute informat=
ion around in a SIP INVITE would accomplish with regards to authorizing CCM=
P or BFCP transactions.

SIP INVITE will carry attributes to authorize that specific SIP request; th=
ere is nothing in proposal #3 that suggest that SIP will carry authorizatio=
n for CCMP or any other protocol.

That is where I think you and I have a disconnect. The SIP INVITE in the co=
nference control (RFC4579) case would not carry attributes like the ones yo=
u've identified in order to authorize that specific SIP request. The attrib=
utes that you identified - the media types the conference allows and the nu=
mber of participants permitted simultaneously in a conference - have nothin=
g to do with the SIP INVITE that opens the conference bridge. Instead, they=
 are components of conference policy. Those policies apply to authorization=
 decisions regarding later participants that will be added to the conferenc=
e, and thus to the authorization of later SIP transactions and dialogs, suc=
h as when a new participant sends an INVITE to the conference bridge after =
having been sent a REFER by the conference controller, say, and the focus d=
ecides to reject that INVITE because the maximum number of participants has=
 been exceeded.

We can imagine a way to publish such a conference policy to the conference =
bridge with SIP, of course, but the question is, in what sense are those po=
licies actually aligned with the question of whether or not we should autho=
rize the SIP request that carries them? If they are not, then we aren't rea=
lly talking about authorizing SIP requests, we're talking uploading long-te=
rm application policies over SIP with faux-HTTP semantics, which is pretty =
far from where we started and where I think we need to go.


So again, the point of my last mail was that the problem of authorizing SIP=
 INVITEs is actually different from the problem of authorizing generic appl=
ication transactions. SIP INVITEs still have more or less the semantics of =
a call setup message: they are requesting that a session be initiated. Conf=
erence control is not managed with SIP transactions, and the security of co=
nference control protocols does not devolve to authorizing SIP transactions=
.

Well, RFC4579 defines a SIP Conference Control mechanism that allows you to=
 initiate and control a conference server using only SIP requests; so I am =
not sure I understand this statement, unless we have different definitions =
for the term =93conference control=94.

Yeah, I was using the term loosely - once we bring RFC4579 into the mix, we=
 need to distinguish "conference control" from "conference policy," and the=
n say that RFC4579 and the entire RFC4353 framework is about control, not a=
bout managing conference policy over SIP. The two attributes you propose be=
long to that conference policy realm rather than conference control, strict=
ly speaking.
In any case, my point was not to discuss conference servers specifically, b=
ut to discuss the general idea of delegating both AuthN and AuthZ in a SIP =
network.

In that case, if you could furnish a use case with attributes that would be=
 carried in a SIP iNVITE that are actually salient to the authorization of =
that particular SIP request, we'd be getting a lot closer to common ground.

Jon Peterson
Neustar, Inc.

Regards,
 Rifaat



I suspect we'll find that most of the other security attributes that "appli=
cations" need to do their jobs similarly don't map onto the semantics of a =
SIP INVITE - but, I'm still open to entertaining examples that might cast d=
oubt on that suspicion.

Jon Peterson
Neustar, Inc.

From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com<mailto:rifaat.ietf@gmail.co=
m>>
Date: Thursday, June 9, 2016 at 10:59 AM
To: Jon Peterson <jon.peterson@neustar.biz<mailto:jon.peterson@neustar.biz>=
>
Cc: Christer Holmberg <christer.holmberg@ericsson.com<mailto:christer.holmb=
erg@ericsson.com>>, "sipcore@ietf.org<mailto:sipcore@ietf.org>" <sipcore@ie=
tf.org<mailto:sipcore@ietf.org>>
Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3

Hi Jon,

Sorry for the delayed reply.

Let's take the video conference example, and discuss the possible ways for =
the conference server to provide service.


         Trust Domain 1              :    Internet     :  Trust Domain 2
                                     :                 :
                      +--------+     :                 :
                      | IdP    |     :                 :
        +-------------| AuthZ  |     :                 :
        |             |        |     :                 :
        |             +--------+     :                 :
        |                 |          :                 :
        |                 |          :                 :
    +--------+        +--------+     :                 :     +--------+
    |        |        | SIP    |     :                 :     | SIP App|
    |   UA   |--------| Proxy  |-----:-----------------:-----| Video  |
    |        |        |        |     :                 :     | Conference
    +--------+        +--------+     :                 :     +--------+
                                     :                 :

Possible solutions:

1. Conference Server acts as an AuthN and AuthZ server. Requests to the ser=
ver must provide an identity and credentials, and services will be provided=
 based on locally stored information associated with the identity provided =
in the request.

2. Conference Server acts as an AuthZ server, but delegates the AuthN part.=
 Requests to the server must provide an identity token, and services will b=
e provided based on locally stored information associated with the identity=
 provided in the token.

3. Conference Server delegates both AuthN and AuthZ parts. Requests to the =
server must provide an access token, which might or might not include infor=
mation about the user requesting the service, and services will be provided=
 based on information provided in the access token or as a result of an int=
rospection step using the access token.


Now, let's look at the 3rd option a bit more, and assume that the server pr=
ovides the following types of conference services:
1. audio conference
2. audio/video conference
3. web conference

or some combination of the above.

Let's also assume that the server allows control over the number of partici=
pants allowed in each one of these conference types or combinations.


Here is one way that I envisioned we could use to address this use case:

When an enterprise user in the 1st trust domain wants to register with the =
SIP Proxy, it will have to authenticate to the IdP and as a result either t=
he UA or the Proxy (depends on the UA's capabilities) will be provided with=
 an AuthN and AuthZ tokens. This process could involve an interim step to a=
llow only the proxy to get access to the tokens, or the tokens could be sen=
t to the UA which will then use them to communicate with the proxy.

The AuthN token is the identity of the user, and the AuthZ token has all th=
e services granted to the user by the IdP/AuthZ Service.

When the enterprise user needs access to the conference service, the UA wil=
l send an INVITE to the Proxy, and will include the AuthZ token if it is in=
 possession of the tokens; otherwise, the proxy will add the appropriate to=
ken to the outgoing INVITE sent to the conference server.

The scope of the AuthZ token might grant the user access to multiple servic=
es; to avoid sending that AuthZ token to all servers providing services gra=
nted to the user, the proxy could obtain a different token with limited sco=
pe that is specific to the service being provided (conference services in t=
his case); there is a standard Token Exchange mechanism for that.

The AuthZ token will include information about the type of service allowed =
(audio, audio/video, etc), and the limit on the number of participants. The=
 AuthZ token might or might not have any information about the user request=
ing the service, which depends on the level of privacy needed.

When the conference server receives this token in the SIP INVITE, all it ha=
s to do is validate that the token was signed by an IdP that it has a trust=
 relationship with, and provide the service requested by the AuthZ token. T=
he conference server does not need the AuthN token to be able to provide a =
service.

Regards,
 Rifaat


On Wed, May 25, 2016 at 3:06 PM, Peterson, Jon <jon.peterson@neustar.biz<ma=
ilto:jon.peterson@neustar.biz>> wrote:

Also, if we decided to separate the AuthN and AuthZ, then we might need to =
carry more than one token.

I could see an argument that there is a need to carry the identity of the u=
ser in one header, and a reference to a place where you can get attributes =
about the originating user in another header, say. That's more or less what=
 we envisioned the last time we went down this path, which led to the RFC44=
84 requirements, largely informed by SAML (which I seem to recall 3GPP was =
interested in at the time).

But it's no accident that the work on RFC4484 never went beyond requirement=
s. Today, I think that much of the useful work that could be done on the au=
thorization question is in identifying which SIP authorization policies are=
 likely to depend on attributes as opposed to just the identity of the end =
user. While we can readily conceive of applications adjunct to communicatio=
ns that would require attributes, the challenge has been justifying why you=
 would invoke those applications with SIP rather than, say, HTTP.  The fund=
amental semantics of a SIP INVITE have remained pretty close to those of pl=
acing a telephone call.  The authorization policies of a telephone call are=
 really different than that of a web service, say, both in terms of how and=
 why they are invoked, what kinds of parties are involved, what forms redir=
ection or cross-domain interaction can occur. With appropriate caveats for =
other methods (notably MESSAGE, SUBSCRIBE/NOTIFY. INFO), SIP requests have =
authorization policy requirements that look a lot like call treatment polic=
ies.

That much said, I am kind of interested in call treatment policies, and the=
ir interaction with identity information. But just saying that there might =
be a video conference service in the cloud, say, doesn't shed a lot of ligh=
t for me. If there's a video conference, I'd ordinarily think you just call=
 into it. If the user who opens the conference bridge needs to be charged, =
that's identity information - which in its usual form will convey the indiv=
idual user and their domain, like user@domain. If you need more complex sem=
antics than that, it's probably because you're doing something more complic=
ated than just calling in to the conference. But it's less clear to me why =
you'd use SIP for such cases. Drilling down deeper into these use cases wou=
ld help me understand where we're going, anyway - the use cases in RFC4484 =
Section 4, which largely involve billing (including when dealing with vario=
us protocol gateways) and resource management (both preemption and priority=
), ultimately didn't seem sufficient to motivate protocol work.

Another issues in the need, in some cases, to avoid sending the token to th=
e UA; in this case there is a need for an interim step to get the token(s) =
to the SIP proxy.

If this information is carried in headers, then presumably it could be adde=
d, consumed, subtracted, or whatever by any relevant intermediaries.

Jon Peterson
Neustar, Inc.



--_000_D385862C19C10Cjonpetersonneustarbiz_
Content-Type: text/html; charset="Windows-1252"
Content-ID: <F9A7AD451798F64B9A27414D9E13CFFB@neustar.biz>
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</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>This time my response should be more complete. Sorry about that!</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div>
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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>Radhika sent a helpful summary of conference control protocols to the =
list which surveyed mechanisms like CCMP, BFCP, and MRB, protocols which ac=
tually let you manage attributes of a conference like the number of partici=
pants and media types allowed.
</div>
</div>
</blockquote>
<div><br>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">This conference use case was an e=
xample for a SIP-based conference server (RFC4579), just to describe the id=
ea around the delegation of AuthNZ.</font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">If the server is a CCMP-based con=
ference, then you would obviously use that to control and manage the confer=
ence server.</font></p>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>
<div>While RFC4579 (an implementation of the RFC4353 framework) provides a =
means to do conference control from within SIP, the kinds of control primit=
ives it specifies are totally decoupled from the sorts of policies that the=
 attributes you proposed would constrain.
 Those policies fall under what RFC4353 calls &quot;conference policy.&quot=
; Neither RFC4579 nor its requirements (RFC4245) really say anything about =
how conference policy is set or distributed. RFC4353 does, though, and what=
 it says is (in 4.6) is &quot;the conference policy
 can be manipulated using web applications or voice applications [i.e. IVRs=
 - JFP]. It can also be manipulated with non-SIP-specific standard or propr=
ietary protocols.&quot;</div>
<div><br>
</div>
<div>Authorizing SIP requests is misaligned with setting conference policy,=
 especially in consideration of the sorts of attributes you seem to have in=
 mind - even when we're considering controlling conferences compliant with =
the RFC4353 framework. &nbsp;That's because
 the conference in RFC4579 isn't &quot;created&quot; when a user agent send=
s an INVITE request to the conference bridge - it is created when the confe=
rence URI is created, which is necessarily beforehand (or at best, ad hoc t=
hrough the conference factory). Conference
 policy is set and disseminated in that context, not in the context of the =
SIP transaction that happens to open the bridge.</div>
<div><br>
</div>
<div>Moreover, my point was that the example of CCMP shows that you can do =
conference control with OAuth over in the HTTP realm without needing to add=
 anything whatsoever to SIP. &nbsp;If you need OAuth-style controls for con=
ferencing, that would seem to be a quicker
 path to implementation.</div>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
&nbsp;</p>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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>The interesting thing about protocols like CCMP is that they are not, =
you know, actually transported by SIP. CCMP is transported by HTTP. Why? Be=
cause the semantics of conference control doesn't actually map onto SIP INV=
ITE transactions very well. They
 map quite aptly to HTTP, and in the HTTP context, you can certainly use OA=
uth if you like.</div>
</div>
</blockquote>
<div>&nbsp;</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">Help me understand this part.</fo=
nt></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">Why can an HTTP request transport=
 a token to a resource server to be able to access a specific resource, whi=
le SIP request cannot transport a similar token to a SIP server to access a=
 service?</font></p>
</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>
<div>Of course both SIP and HTTP can carry arbitrary chunks of data, includ=
ing any sort of token we can imagine. My point is that the semantics of an =
INVITE are really different from the semantics of a POST - and that is why,=
 ultimately, protocols like CCMP
 don't use SIP as a transport but instead use HTTP. &nbsp;INVITEs are sent =
by a user agent when it is trying to set up a media session between itself =
and some remote resource on the Internet. While RFC4579 has a way to open a=
 conference as a side effect of sending
 an INVITE to connect to the conference, the policies associated with the s=
orts of attributes you specified are set totally outside the context of tha=
t dialog, and the execution of those policies is completely disconnected fr=
om the authorization of the conference-opening
 INVITE. More on this below.</div>
<div><br>
</div>
<div>&nbsp;I think the interesting question about standardizing an approach=
 to authorization of SIP requests is about INVITE. Of course, SIP supports =
faux-HTTP semantics with mechanisms like PUBLISH and NOTIFY, and we could t=
alk about authorizing a number of other
 non-dialog-forming methods as well, but if we're going to kick off an effo=
rt to specify authorization policy for SIP, the primary thing we need to wo=
rry about is INVITE.</div>
</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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>You even will find some text about Role-Based Access Control in the XC=
ON (RFC6503) security consideration. Although you can use SIP and SDP to se=
t up a BFCP connection (RFC4583), the security of BFCP relies on mutual TLS=
 between its own endpoints. It is
 unclear to me what carrying attribute information around in a SIP INVITE w=
ould accomplish with regards to authorizing CCMP or BFCP transactions.</div=
>
</div>
</blockquote>
<div><br>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">SIP INVITE will carry attributes =
to authorize that specific SIP request; there is nothing in proposal #3 tha=
t suggest that SIP will carry authorization for CCMP or any other protocol.=
</font></p>
</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>
<div>That is where I think you and I have a disconnect. The SIP INVITE in t=
he conference control (RFC4579) case would not carry attributes like the on=
es you've identified in order to authorize that specific SIP request. The a=
ttributes that you identified -
 the media types the conference allows and the number of participants permi=
tted simultaneously in a conference - have nothing to do with the SIP INVIT=
E that opens the conference bridge. Instead, they are components of confere=
nce policy. Those policies apply
 to authorization decisions regarding later participants that will be added=
 to the conference, and thus to the authorization of later SIP transactions=
 and dialogs, such as when a new participant sends an INVITE to the confere=
nce bridge after having been sent
 a REFER by the conference controller, say, and the focus decides to reject=
 that INVITE because the maximum number of participants has been exceeded.<=
/div>
<div><br>
</div>
<div>We can imagine a way to publish such a conference policy to the confer=
ence bridge with SIP, of course, but the question is, in what sense are tho=
se policies actually aligned with the question of whether or not we should =
authorize the SIP request that carries
 them? If they are not, then we aren't really talking about authorizing SIP=
 requests, we're talking uploading long-term application policies over SIP =
with faux-HTTP semantics, which is pretty far from where we started and whe=
re I think we need to go.</div>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<span style=3D"font-family:Arial,sans-serif;font-size:12pt"></span></p>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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>So again, the point of my last mail was that the problem of authorizin=
g SIP INVITEs is actually different from the problem of authorizing generic=
 application transactions. SIP INVITEs still have more or less the semantic=
s of a call setup message: they
 are requesting that a session be initiated. Conference control is not mana=
ged with SIP transactions, and the security of conference control protocols=
 does not devolve to authorizing SIP transactions.
</div>
</div>
</blockquote>
<div><br>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">Well, RFC4579 defines a SIP Confe=
rence Control mechanism that allows you to initiate and control a conferenc=
e server using only SIP requests; so I am not sure I understand this statem=
ent, unless we have different definitions
 for the term =93conference control=94.</font></p>
</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>Yeah, I was using the term loosely - once we bring RFC4579 into the mi=
x, we need to distinguish &quot;conference control&quot; from &quot;confere=
nce policy,&quot; and then say that RFC4579 and the entire RFC4353 framewor=
k is about control, not about managing conference policy
 over SIP. The two attributes you propose belong to that conference policy =
realm rather than conference control, strictly speaking.</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<span style=3D"font-family: arial, helvetica, sans-serif;">In any case, my =
point was not to discuss conference servers specifically, but to discuss th=
e general idea of delegating both AuthN and AuthZ in a SIP network.</span><=
/p>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>
<div>In that case, if you could furnish a use case with attributes that wou=
ld be carried in a SIP iNVITE that are actually salient to the authorizatio=
n of that particular SIP request, we'd be getting a lot closer to common gr=
ound.</div>
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">&nbsp;</font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">Regards,</font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">&nbsp;Rifaat</font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<span style=3D"font-size:12pt;font-family:Arial,sans-serif"><br>
</span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<span style=3D"font-size:12pt;font-family:Arial,sans-serif"><br>
</span></p>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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 suspect we'll find that most of the other security attributes that &=
quot;applications&quot; need to do their jobs similarly don't map onto the =
semantics of a SIP INVITE - but, I'm still open to entertaining examples th=
at might cast doubt on that suspicion.&nbsp;</div>
<div><br>
</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, June 9, 2016 at 10:=
59 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>Christer Holmberg &lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmb=
erg@ericsson.com</a>&gt;, &quot;<a href=3D"mailto:sipcore@ietf.org" target=
=3D"_blank">sipcore@ietf.org</a>&quot; &lt;<a href=3D"mailto:sipcore@ietf.o=
rg" target=3D"_blank">sipcore@ietf.org</a>&gt;<span class=3D""><br>
<span style=3D"font-weight:bold">Subject: </span>Re: [sipcore] SIP AuthNZ P=
roblem Statement - v3<br>
</span></div>
<div>
<div class=3D"h5">
<div><br>
</div>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div>
<div>
<div dir=3D"ltr"><font face=3D"monospace,monospace">Hi Jon,</font>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Sorry for the delayed reply.</font>=
</div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div>
<div>
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><font face=3D"monospace,monospace">Let's take the video conference exa=
mple, and discuss the possible ways for&nbsp;</font><span style=3D"font-fam=
ily:monospace,monospace">the conference server to provide service.</span></=
div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;T=
rust Domain 1 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;: &nbsp; &nbs=
p;Internet &nbsp; &nbsp; : &nbsp;Trust Domain 2</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &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; :</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &#43;--------&#43; &nbsp; &nbsp; =
: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | IdP &nbsp; &nbsp;| &nbsp; &nbsp=
; : &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; &#43;--=
-----------| AuthZ &nbsp;| &nbsp; &nbsp; : &nbsp; &nbsp; &nbsp; &nbsp; &nbs=
p; &nbsp; &nbsp; &nbsp; :</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; | &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; =
&nbsp; : &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</font></=
div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; | &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &#43;--------&#43; &nbsp; &nbsp; : &nb=
sp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; | &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &=
nbsp; &nbsp;: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</fo=
nt></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &nbsp; &nbsp; | &nbsp=
; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &=
nbsp; &nbsp;: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; :</fo=
nt></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &#43;--------&#43; &n=
bsp; &nbsp; &nbsp; &nbsp;&#43;--------&#43; &nbsp; &nbsp; : &nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : &nbsp; &nbsp; &#43;--------&#43;=
</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; | &nbsp; &nbsp; &nbsp=
; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp;| SIP &nbsp; &nbsp;| &nbsp; &nbsp; : &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : &nbsp; &nbsp; | SI=
P App|</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; | &nbsp; UA &nbsp; |-=
-------| Proxy &nbsp;|-----:-----------------:-----| Video &nbsp;|</font></=
div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; | &nbsp; &nbsp; &nbsp=
; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; =
&nbsp; : &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : &nbsp; &=
nbsp; | Conference</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &#43;--------&#43; &n=
bsp; &nbsp; &nbsp; &nbsp;&#43;--------&#43; &nbsp; &nbsp; : &nbsp; &nbsp; &=
nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : &nbsp; &nbsp; &#43;--------&#43;=
</font></div>
<div><font face=3D"monospace,monospace">&nbsp; &nbsp; &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; :</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Possible solutions:</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">1. Conference Server acts as an Aut=
hN and AuthZ server. Requests to the&nbsp;server must provide an identity a=
nd credentials, and services will be&nbsp;provided based on locally stored =
information associated with the&nbsp;identity provided
 in the request.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">2. Conference Server acts as an Aut=
hZ server, but delegates the AuthN part.&nbsp;Requests to the server must p=
rovide an identity token, and services will&nbsp;be provided based on local=
ly stored information associated with the&nbsp;identity
 provided in the token.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">3. Conference Server delegates both=
 AuthN and AuthZ parts. Requests to the&nbsp;server must provide an access =
token, which might or might not include&nbsp;information about the user req=
uesting the service, and services will&nbsp;be provided
 based on information provided in the access token or as a&nbsp;result of a=
n introspection step using the access token.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Now, let's look at the 3rd option a=
 bit more, and assume that the server&nbsp;provides the following types of =
conference services:</font></div>
<div><font face=3D"monospace,monospace">1. audio conference</font></div>
<div><font face=3D"monospace,monospace">2. audio/video conference</font></d=
iv>
<div><font face=3D"monospace,monospace">3. web conference</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">or some&nbsp;combination&nbsp;of th=
e above.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Let's also assume that the server a=
llows control over the number of&nbsp;participants allowed in each one of t=
hese conference types or combinations.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Here is one way that I envisioned w=
e could use to address this use case:</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
</div>
</div>
</div>
<font face=3D"monospace,monospace">When an enterprise user in the 1st trust=
 domain wants to register with the&nbsp;SIP Proxy, it will have to authenti=
cate to the IdP and as a result either&nbsp;the UA or the Proxy (depends on=
 the UA's capabilities) will be provided&nbsp;with
 an AuthN and AuthZ tokens. This process could involve an interim step&nbsp=
;to allow only the proxy to get access to the tokens, or the tokens could b=
e&nbsp;sent to the UA which will then use them to communicate with the prox=
y.</font>
<div><font face=3D"monospace,monospace"><br>
The AuthN token is the identity of the user, and the AuthZ token has all&nb=
sp;the services granted to the user by the IdP/AuthZ Service.<br>
<br>
When the enterprise user needs access to the conference service, the UA&nbs=
p;will send an INVITE to the Proxy, and will include the AuthZ token if it =
is&nbsp;in possession of the tokens; otherwise, the proxy will add the appr=
opriate&nbsp;token to the outgoing INVITE sent
 to the conference server.<br>
<br>
The scope of the AuthZ token might grant the user access to multiple&nbsp;s=
ervices; to avoid sending that AuthZ token to all servers providing&nbsp;se=
rvices granted to the user, the proxy could obtain a different token with&n=
bsp;limited scope that is specific to the service
 being provided (conference&nbsp;services in this case); there is a standar=
d Token Exchange mechanism for&nbsp;that.<br>
<br>
The AuthZ token will include information about the type of service allowed&=
nbsp;(audio, audio/video, etc), and the limit on the number of participants=
.&nbsp;The AuthZ token might or might not have any information about the us=
er&nbsp;requesting the service, which depends on
 the level of privacy needed.<br>
<br>
When the conference server receives this token in the SIP INVITE, all it&nb=
sp;has to do is validate that the token was signed by an IdP that it has a&=
nbsp;trust relationship with, and provide the service requested by the Auth=
Z&nbsp;token. The conference server does not need
 the AuthN token to be able to&nbsp;provide a service.</font>
<div>
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Regards,</font></div>
<div><font face=3D"monospace,monospace">&nbsp;Rifaat</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
</div>
</div>
</div>
</div>
</div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Wed, May 25, 2016 at 3:06 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-style:solid;border-left-color:rgb(204,204,204);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">
<span><span>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
</div>
</div>
</div>
</span><span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>Also, if we decided to separate the AuthN and AuthZ, then we might nee=
d to carry more than one token.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span>
<div>I could see an argument that there is a need to carry the identity of =
the user in one header, and a reference to a place where you can get attrib=
utes about the originating user in another header, say. That's more or less=
 what we envisioned the last time
 we went down this path, which led to the RFC4484 requirements, largely inf=
ormed by SAML (which I seem to recall 3GPP was interested in at the time).<=
/div>
<div><br>
</div>
<div>But it's no accident that the work on RFC4484 never went beyond requir=
ements. Today, I think that much of the useful work that could be done on t=
he authorization question is in identifying which SIP authorization policie=
s are likely to depend on attributes
 as opposed to just the identity of the end user. While we can readily conc=
eive of applications adjunct to communications that would require attribute=
s, the challenge has been justifying why you would invoke those application=
s with SIP rather than, say, HTTP.
 &nbsp;The fundamental semantics of a SIP INVITE have remained pretty close=
 to those of placing a telephone call.&nbsp; The authorization policies of =
a telephone call are really different than that of a web service, say, both=
 in terms of how and why they are invoked,
 what kinds of parties are involved, what forms redirection or cross-domain=
 interaction can occur. With appropriate caveats for other methods (notably=
 MESSAGE, SUBSCRIBE/NOTIFY. INFO), SIP requests have authorization policy r=
equirements that look a lot like
 call treatment policies.</div>
<div><br>
</div>
<div>That much said, I am kind of interested in call treatment policies, an=
d their interaction with identity information. But just saying that there m=
ight be a video conference service in the cloud, say, doesn't shed a lot of=
 light for me. If there's a video
 conference, I'd ordinarily think you just call into it. If the user who op=
ens the conference bridge needs to be charged, that's identity information =
- which in its usual form will convey the individual user and their domain,=
 like user@domain. If you need more
 complex semantics than that, it's probably because you're doing something =
more complicated than just calling in to the conference. But it's less clea=
r to me why you'd use SIP for such cases. Drilling down deeper into these u=
se cases would help me understand
 where we're going, anyway - the use cases in RFC4484 Section 4, which larg=
ely involve billing (including when dealing with various protocol gateways)=
 and resource management (both preemption and priority), ultimately didn't =
seem sufficient to motivate protocol
 work.</div>
<span>
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>Another issues in the need, in some cases, to avoid sending the token =
to the UA; in this case there is a need for an interim step to get the toke=
n(s) to the SIP proxy.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span>
<div>If this information is carried in headers, then presumably it could be=
 added, consumed, subtracted, or whatever by any relevant intermediaries.&n=
bsp;</div>
<span><font color=3D"#888888">
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
</font></span></div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</blockquote>
</div>
</div>
</span></div>
</blockquote>
</div>
<br>
</div>
</div>
</blockquote>
</span>
</body>
</html>

--_000_D385862C19C10Cjonpetersonneustarbiz_--


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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "Olle E. Johansson" <oej@edvina.net>
Thread-Topic: [sipcore] SIPS must die, die, die
Thread-Index: AQHRw2RtJJyRZYItSEGGroXfIB854p/kFpuAgACq/ACAAPxBgIAB9SyA///VhwCAAVK9gIAAMDEA
Date: Tue, 14 Jun 2016 17:29:24 +0000
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Subject: Re: [sipcore] SIPS must die, die, die
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Just to add a bit to what Brian said...

>>=20
>I think we need to start with =B3what if we remove SIPS=B2 and go back and
>see what=B9s missing and how we can fix that. A lot of people point to SIP=
S
>as
>the solution - but no one is implementing or using it.

Ostensibly what would be missing is a mechanism for user agents to
indicate that any intermediaries handling their request should only
forward it with hop-by-hop confidentiality and integrity protections. And
for SIP resources to indicate their preference to be reached in this
fashion.

>=20
>What is the problem it solves?

Pervasive eavesdropping?

> Is there another way?

Maybe.

>What=B9s the status for S/MIME.
>
>I don=B9t think we=B9ll find that the answer is SIPS: as a URI scheme, but=
 the
>SIP over TLS transport is still very much needed, discovered by NAPTR/SRV
>and supported by DANE or some other PKI structure.

If your intention is that we should replace SIPS with something that
provides a stronger assurance of hop-by-hop signaling confidentiality,
then great, let's specify that stronger mechanism first and then we'll
deprecate SIPS in favor of it. If the process is instead, let's deprecate
SIPS and then hope we can invent something else to address pervasive
eavesdropping later, I would probably say that's unacceptable, provided we
think there are any deployment environments in which SIPS could work as
advertised. And yes, I'm aware there are deployment environments in which
it is just a nuisance, but coincidentally, those are also deployment
environments that are not overly concerned with pervasive eavesdropping.

Again, discovering via NAPTR/SRV that the resource wants to be reached
over TLS was one of the primary functions for which SIPS was devised. I'm
not sure how different anything we come up along those lines would
realistically be from what's already in RFC3263.

Jon Peterson
Neustar, Inc.


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Peterson, Jon" <jon.peterson@neustar.biz>, Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Thread-Topic: [sipcore] SIP AuthNZ Problem Statement - v3
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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Hi,

.

>> Why can an HTTP request transport a token to a resource server to be abl=
e to access a specific resource, while SIP request cannot transport a simil=
ar token to a SIP server to access a service?
>
> Of course both SIP and HTTP can carry arbitrary chunks of data, including=
 any sort of token we can imagine. My point is that the semantics of an INV=
ITE are really different from the semantics of a POST - > and that is why, =
ultimately, protocols like CCMP don't use SIP as a transport but instead us=
e HTTP. =A0INVITEs are sent by a user agent when it is trying to set up a m=
edia session between itself and some=20
> remote resource on the Internet. While RFC4579 has a way to set up a conf=
erence as a side effect of sending an INVITE to connect to the conference, =
must of the subsequent control work happens=20
> totally outside the context of that dialog, and the authorization of thos=
e operations is completely disconnected from the authorization of the confe=
rence-opening INVITE.
>
> And of course, SIP can support faux-HTTP semantics with mechanisms like P=
UBLISH and NOTIFY. But I think the interesting question about standardizing=
 an approach to authorization of SIP requests is=20
> about INVITE.

Just to point out that in the IMS use-case the token needs to be carried in=
 the REGISTER request, and it has nothing to do with a conference server.

Regards,

Christer



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Subject: Re: [sipcore] SIPS must die, die, die
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Thread-Topic: [sipcore] SIP AuthNZ Problem Statement - v3
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Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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But to the point - in that (very very different) use case, the REGISTER
request is accepted or rejected based on the contents of the token it
carries, right?

Jon Peterson
Neustar, Inc.

On 6/14/16, 10:42 AM, "Christer Holmberg" <christer.holmberg@ericsson.com>
wrote:

>Hi,
>
>.
>
>>> Why can an HTTP request transport a token to a resource server to be
>>>able to access a specific resource, while SIP request cannot transport
>>>a similar token to a SIP server to access a service?
>>
>> Of course both SIP and HTTP can carry arbitrary chunks of data,
>>including any sort of token we can imagine. My point is that the
>>semantics of an INVITE are really different from the semantics of a POST
>>- > and that is why, ultimately, protocols like CCMP don't use SIP as a
>>transport but instead use HTTP.  INVITEs are sent by a user agent when
>>it is trying to set up a media session between itself and some
>> remote resource on the Internet. While RFC4579 has a way to set up a
>>conference as a side effect of sending an INVITE to connect to the
>>conference, must of the subsequent control work happens
>> totally outside the context of that dialog, and the authorization of
>>those operations is completely disconnected from the authorization of
>>the conference-opening INVITE.
>>
>> And of course, SIP can support faux-HTTP semantics with mechanisms like
>>PUBLISH and NOTIFY. But I think the interesting question about
>>standardizing an approach to authorization of SIP requests is
>> about INVITE.
>
>Just to point out that in the IMS use-case the token needs to be carried
>in the REGISTER request, and it has nothing to do with a conference
>server.
>
>Regards,
>
>Christer
>
>


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Peterson, Jon" <jon.peterson@neustar.biz>, Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
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Hi,

>But to the point - in that (very very different) use case, the REGISTER re=
quest is accepted or rejected >based on the contents of the token it carrie=
s, right?

Yes.

Regards,

Christer


On 6/14/16, 10:42 AM, "Christer Holmberg" <christer.holmberg@ericsson.com>
wrote:

>Hi,
>
>.
>
>>> Why can an HTTP request transport a token to a resource server to be=20
>>>able to access a specific resource, while SIP request cannot=20
>>>transport a similar token to a SIP server to access a service?
>>
>> Of course both SIP and HTTP can carry arbitrary chunks of data,=20
>>including any sort of token we can imagine. My point is that the=20
>>semantics of an INVITE are really different from the semantics of a=20
>>POST
>>- > and that is why, ultimately, protocols like CCMP don't use SIP as=20
>>a transport but instead use HTTP.  INVITEs are sent by a user agent=20
>>when it is trying to set up a media session between itself and some =20
>>remote resource on the Internet. While RFC4579 has a way to set up a=20
>>conference as a side effect of sending an INVITE to connect to the=20
>>conference, must of the subsequent control work happens  totally=20
>>outside the context of that dialog, and the authorization of those=20
>>operations is completely disconnected from the authorization of the=20
>>conference-opening INVITE.
>>
>> And of course, SIP can support faux-HTTP semantics with mechanisms=20
>>like PUBLISH and NOTIFY. But I think the interesting question about=20
>>standardizing an approach to authorization of SIP requests is  about=20
>>INVITE.
>
>Just to point out that in the IMS use-case the token needs to be=20
>carried in the REGISTER request, and it has nothing to do with a=20
>conference server.
>
>Regards,
>
>Christer
>
>


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To: "Peterson, Jon" <jon.peterson@neustar.biz>, "Olle E. Johansson" <oej@edvina.net>
References: <9C858D97-7508-483A-8823-507F7F10ADDA@edvina.net> <87d1no3dgz.fsf@hobgoblin.ariadne.com> <7594FB04B1934943A5C02806D1A2204B3804B3B5@ESESSMB209.ericsson.se> <c680f61f-cb3b-f607-2529-715fb8e99119@alum.mit.edu> <46229759-0887-4C3F-B84D-51BA591B1ED6@att.com> <949EF20990823C4C85C18D59AA11AD8BADF14B42@FR712WXCHMBA11.zeu.alcatel-lucent.com> <D38429C2.19BA6C%jon.peterson@neustar.biz> <492A42E2-FD69-4882-B1DA-E0A9277D7125@edvina.net> <D38569DC.19BE1C%jon.peterson@neustar.biz>
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] SIPS must die, die, die
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On 6/14/16 1:29 PM, Peterson, Jon wrote:
>
> Just to add a bit to what Brian said...
>
>>>
>> I think we need to start with ³what if we remove SIPS² and go back and
>> see what¹s missing and how we can fix that. A lot of people point to SIPS
>> as
>> the solution - but no one is implementing or using it.
>
> Ostensibly what would be missing is a mechanism for user agents to
> indicate that any intermediaries handling their request should only
> forward it with hop-by-hop confidentiality and integrity protections. And
> for SIP resources to indicate their preference to be reached in this
> fashion.

What is the point of the hop-by-hop protections in this context? Do we 
really need it? Won't STIR and SIPBRANDY provide what is needed without it?

>> What is the problem it solves?
>
> Pervasive eavesdropping?

Since hop-by-hop presumes that all the intermediaries are trusted, and 
this rarely seems like a good assumption for preventing pervasive 
eavesdropping, maybe it isn't worth the trouble.

	Thanks,
	Paul

>> Is there another way?
>
> Maybe.
>
>> What¹s the status for S/MIME.
>>
>> I don¹t think we¹ll find that the answer is SIPS: as a URI scheme, but the
>> SIP over TLS transport is still very much needed, discovered by NAPTR/SRV
>> and supported by DANE or some other PKI structure.
>
> If your intention is that we should replace SIPS with something that
> provides a stronger assurance of hop-by-hop signaling confidentiality,
> then great, let's specify that stronger mechanism first and then we'll
> deprecate SIPS in favor of it. If the process is instead, let's deprecate
> SIPS and then hope we can invent something else to address pervasive
> eavesdropping later, I would probably say that's unacceptable, provided we
> think there are any deployment environments in which SIPS could work as
> advertised. And yes, I'm aware there are deployment environments in which
> it is just a nuisance, but coincidentally, those are also deployment
> environments that are not overly concerned with pervasive eavesdropping.
>
> Again, discovering via NAPTR/SRV that the resource wants to be reached
> over TLS was one of the primary functions for which SIPS was devised. I'm
> not sure how different anything we come up along those lines would
> realistically be from what's already in RFC3263.
>
> Jon Peterson
> Neustar, Inc.
>
>


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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "Olle E. Johansson" <oej@edvina.net>
Thread-Topic: [sipcore] SIPS must die, die, die
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Date: Tue, 14 Jun 2016 20:45:22 +0000
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References: <9C858D97-7508-483A-8823-507F7F10ADDA@edvina.net> <87d1no3dgz.fsf@hobgoblin.ariadne.com> <7594FB04B1934943A5C02806D1A2204B3804B3B5@ESESSMB209.ericsson.se> <c680f61f-cb3b-f607-2529-715fb8e99119@alum.mit.edu> <46229759-0887-4C3F-B84D-51BA591B1ED6@att.com> <949EF20990823C4C85C18D59AA11AD8BADF14B42@FR712WXCHMBA11.zeu.alcatel-lucent.com> <D38429C2.19BA6C%jon.peterson@neustar.biz> <492A42E2-FD69-4882-B1DA-E0A9277D7125@edvina.net> <D38569DC.19BE1C%jon.peterson@neustar.biz> <28e047ef-890c-1c9b-e280-eee39107361b@alum.mit.edu>
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Subject: Re: [sipcore] SIPS must die, die, die
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To: "Peterson, Jon" <jon.peterson@neustar.biz>, "Olle E. Johansson" <oej@edvina.net>
References: <9C858D97-7508-483A-8823-507F7F10ADDA@edvina.net> <87d1no3dgz.fsf@hobgoblin.ariadne.com> <7594FB04B1934943A5C02806D1A2204B3804B3B5@ESESSMB209.ericsson.se> <c680f61f-cb3b-f607-2529-715fb8e99119@alum.mit.edu> <46229759-0887-4C3F-B84D-51BA591B1ED6@att.com> <949EF20990823C4C85C18D59AA11AD8BADF14B42@FR712WXCHMBA11.zeu.alcatel-lucent.com> <D38429C2.19BA6C%jon.peterson@neustar.biz> <492A42E2-FD69-4882-B1DA-E0A9277D7125@edvina.net> <D38569DC.19BE1C%jon.peterson@neustar.biz> <28e047ef-890c-1c9b-e280-eee39107361b@alum.mit.edu> <D385B9BB.19C40A%jon.peterson@neustar.biz>
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Date: Tue, 14 Jun 2016 21:58:45 -0400
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Subject: Re: [sipcore] SIPS must die, die, die
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On 6/14/16 4:45 PM, Peterson, Jon wrote:
>
> The threat of pervasively monitoring signaling is basically metadata
> collection - seeing who called whom at what time, and so on. Neither STIR
> nor SIPBRANDY explicitly addresses that. If the attacker can compromise
> intermediaries, or otherwise persuade them to yield logs of the signaling,
> then agreed SIPS would add little value. But that is far harder to arrange
> than just snooping on cleartext traffic over the wire.
>
> I do think it's a different threat, and it requires different
> countermeasures. Those countermeasures will not be universally available,
> but forcing SIP over TLS as much as possible still would make life harder
> for these passive attackers.

Agreed, but that can be achieved simply by getting TLS to be the default.

That might be an easier thing to sell if we can get SIP over DTLS defined.

	Thanks,
	Paul

> Jon Peterson
> Neustar, Inc.
>
> On 6/14/16, 1:04 PM, "Paul Kyzivat" <pkyzivat@alum.mit.edu> wrote:
>
>> On 6/14/16 1:29 PM, Peterson, Jon wrote:
>>>
>>> Just to add a bit to what Brian said...
>>>
>>>>>
>>>> I think we need to start with ©øwhat if we remove SIPS©÷ and go back and
>>>> see what©ös missing and how we can fix that. A lot of people point to
>>>> SIPS
>>>> as
>>>> the solution - but no one is implementing or using it.
>>>
>>> Ostensibly what would be missing is a mechanism for user agents to
>>> indicate that any intermediaries handling their request should only
>>> forward it with hop-by-hop confidentiality and integrity protections.
>>> And
>>> for SIP resources to indicate their preference to be reached in this
>>> fashion.
>>
>> What is the point of the hop-by-hop protections in this context? Do we
>> really need it? Won't STIR and SIPBRANDY provide what is needed without
>> it?
>>
>>>> What is the problem it solves?
>>>
>>> Pervasive eavesdropping?
>>
>> Since hop-by-hop presumes that all the intermediaries are trusted, and
>> this rarely seems like a good assumption for preventing pervasive
>> eavesdropping, maybe it isn't worth the trouble.
>>
>> 	Thanks,
>> 	Paul
>>
>>>> Is there another way?
>>>
>>> Maybe.
>>>
>>>> What©ös the status for S/MIME.
>>>>
>>>> I don©öt think we©öll find that the answer is SIPS: as a URI scheme, but
>>>> the
>>>> SIP over TLS transport is still very much needed, discovered by
>>>> NAPTR/SRV
>>>> and supported by DANE or some other PKI structure.
>>>
>>> If your intention is that we should replace SIPS with something that
>>> provides a stronger assurance of hop-by-hop signaling confidentiality,
>>> then great, let's specify that stronger mechanism first and then we'll
>>> deprecate SIPS in favor of it. If the process is instead, let's
>>> deprecate
>>> SIPS and then hope we can invent something else to address pervasive
>>> eavesdropping later, I would probably say that's unacceptable, provided
>>> we
>>> think there are any deployment environments in which SIPS could work as
>>> advertised. And yes, I'm aware there are deployment environments in
>>> which
>>> it is just a nuisance, but coincidentally, those are also deployment
>>> environments that are not overly concerned with pervasive eavesdropping.
>>>
>>> Again, discovering via NAPTR/SRV that the resource wants to be reached
>>> over TLS was one of the primary functions for which SIPS was devised.
>>> I'm
>>> not sure how different anything we come up along those lines would
>>> realistically be from what's already in RFC3263.
>>>
>>> Jon Peterson
>>> Neustar, Inc.
>>>
>>>
>>
>


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To: "Peterson, Jon" <jon.peterson@neustar.biz>
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Subject: Re: [sipcore] SIPS must die, die, die
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> On 14 Jun 2016, at 22:45, Peterson, Jon <jon.peterson@neustar.biz> =
wrote:
>=20
>=20
> The threat of pervasively monitoring signaling is basically metadata
> collection - seeing who called whom at what time, and so on. Neither =
STIR
> nor SIPBRANDY explicitly addresses that. If the attacker can =
compromise
> intermediaries, or otherwise persuade them to yield logs of the =
signaling,
> then agreed SIPS would add little value. But that is far harder to =
arrange
> than just snooping on cleartext traffic over the wire.
>=20
> I do think it's a different threat, and it requires different
> countermeasures. Those countermeasures will not be universally =
available,
> but forcing SIP over TLS as much as possible still would make life =
harder
> for these passive attackers.
Thank you Jon!=20

That=E2=80=99s exactly what I want to do by making implementing
TLS in SIP more straightforward, better documented and easier to test.
I do believe the SIPS: uri scheme has been a hindrance and by removing =
it
(or just discussing what will happen if we remove it) we will focus on =
the
right things to make both SIP over TLS in a secure context and SIP over=20=

TLS in opportunistic mode more of a default than it is now.

We=E2=80=99ve come far in the specification of SIPconnect 2.0 (work in =
progress)
but identified a few issues that needs to be solved in the IETF.
One is the =E2=80=9Chalf-outbound=E2=80=9D and another is how to live =
without =E2=80=9C;transport=3Dtls=E2=80=9D.

I=E2=80=99m trying to summarize my view of the discussion about SIPS: =
and some
of the areas I think we need to look at in a separate document.

/O

>=20
> Jon Peterson
> Neustar, Inc.
>=20
> On 6/14/16, 1:04 PM, "Paul Kyzivat" <pkyzivat@alum.mit.edu> wrote:
>=20
>> On 6/14/16 1:29 PM, Peterson, Jon wrote:
>>>=20
>>> Just to add a bit to what Brian said...
>>>=20
>>>>>=20
>>>> I think we need to start with =C2=B3what if we remove SIPS=C2=B2 =
and go back and
>>>> see what=C2=B9s missing and how we can fix that. A lot of people =
point to
>>>> SIPS
>>>> as
>>>> the solution - but no one is implementing or using it.
>>>=20
>>> Ostensibly what would be missing is a mechanism for user agents to
>>> indicate that any intermediaries handling their request should only
>>> forward it with hop-by-hop confidentiality and integrity =
protections.
>>> And
>>> for SIP resources to indicate their preference to be reached in this
>>> fashion.
>>=20
>> What is the point of the hop-by-hop protections in this context? Do =
we
>> really need it? Won't STIR and SIPBRANDY provide what is needed =
without
>> it?
>>=20
>>>> What is the problem it solves?
>>>=20
>>> Pervasive eavesdropping?
>>=20
>> Since hop-by-hop presumes that all the intermediaries are trusted, =
and
>> this rarely seems like a good assumption for preventing pervasive
>> eavesdropping, maybe it isn't worth the trouble.
>>=20
>> 	Thanks,
>> 	Paul
>>=20
>>>> Is there another way?
>>>=20
>>> Maybe.
>>>=20
>>>> What=C2=B9s the status for S/MIME.
>>>>=20
>>>> I don=C2=B9t think we=C2=B9ll find that the answer is SIPS: as a =
URI scheme, but
>>>> the
>>>> SIP over TLS transport is still very much needed, discovered by
>>>> NAPTR/SRV
>>>> and supported by DANE or some other PKI structure.
>>>=20
>>> If your intention is that we should replace SIPS with something that
>>> provides a stronger assurance of hop-by-hop signaling =
confidentiality,
>>> then great, let's specify that stronger mechanism first and then =
we'll
>>> deprecate SIPS in favor of it. If the process is instead, let's
>>> deprecate
>>> SIPS and then hope we can invent something else to address pervasive
>>> eavesdropping later, I would probably say that's unacceptable, =
provided
>>> we
>>> think there are any deployment environments in which SIPS could work =
as
>>> advertised. And yes, I'm aware there are deployment environments in
>>> which
>>> it is just a nuisance, but coincidentally, those are also deployment
>>> environments that are not overly concerned with pervasive =
eavesdropping.
>>>=20
>>> Again, discovering via NAPTR/SRV that the resource wants to be =
reached
>>> over TLS was one of the primary functions for which SIPS was =
devised.
>>> I'm
>>> not sure how different anything we come up along those lines would
>>> realistically be from what's already in RFC3263.
>>>=20
>>> Jon Peterson
>>> Neustar, Inc.
>>>=20
>>>=20
>>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Date: Wed, 15 Jun 2016 10:51:34 -0400
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To: "Peterson, Jon" <jon.peterson@neustar.biz>
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Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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On Tue, Jun 14, 2016 at 1:27 PM, Peterson, Jon <jon.peterson@neustar.biz>
wrote:

>
> This time my response should be more complete. Sorry about that!
>
>
>> Radhika sent a helpful summary of conference control protocols to the
>> list which surveyed mechanisms like CCMP, BFCP, and MRB, protocols which
>> actually let you manage attributes of a conference like the number of
>> participants and media types allowed.
>>
>
> This conference use case was an example for a SIP-based conference server
> (RFC4579), just to describe the idea around the delegation of AuthNZ.
>
> If the server is a CCMP-based conference, then you would obviously use
> that to control and manage the conference server.
>
>
> While RFC4579 (an implementation of the RFC4353 framework) provides a
> means to do conference control from within SIP, the kinds of control
> primitives it specifies are totally decoupled from the sorts of policies
> that the attributes you proposed would constrain. Those policies fall und=
er
> what RFC4353 calls "conference policy."
>

I agree. This is an important distinction and clarification.



> Neither RFC4579 nor its requirements (RFC4245) really say anything about
> how conference policy is set or distributed. RFC4353 does, though, and wh=
at
> it says is (in 4.6) is "the conference policy can be manipulated using we=
b
> applications or voice applications [i.e. IVRs - JFP]. It can also be
> manipulated with non-SIP-specific standard or proprietary protocols."
>
>
Proposal #3 delegates this function from the conference server to the
authorization server. The conference server becomes a consumer and enforcer
of a policy.



> Authorizing SIP requests is misaligned with setting conference policy,
> especially in consideration of the sorts of attributes you seem to have i=
n
> mind - even when we're considering controlling conferences compliant with
> the RFC4353 framework.  That's because the conference in RFC4579 isn't
> "created" when a user agent sends an INVITE request to the conference
> bridge - it is created when the conference URI is created, which is
> necessarily beforehand (or at best, ad hoc through the conference factory=
).
> Conference policy is set and disseminated in that context, not in the
> context of the SIP transaction that happens to open the bridge.
>

In the diagram I provided, I did not separate the Conference Factory from
the Focus, maybe I should have.
The INVITE with the token will be sent to the factory, which will create
the focus and set the proper limitation on the focus based on the
authorization provided in the token.
How is this misaligned with settings a conference policy?



>
> Moreover, my point was that the example of CCMP shows that you can do
> conference control with OAuth over in the HTTP realm without needing to a=
dd
> anything whatsoever to SIP.  If you need OAuth-style controls for
> conferencing, that would seem to be a quicker path to implementation.
>

If your conference server supports CCMP, then yes, that would be the right
answer. What if your server does not support CCMP?
But, again, the conference use case is an example only to drive the point,
not to discuss the best way to implement conference server.



>
>
>> The interesting thing about protocols like CCMP is that they are not, yo=
u
>> know, actually transported by SIP. CCMP is transported by HTTP. Why?
>> Because the semantics of conference control doesn't actually map onto SI=
P
>> INVITE transactions very well. They map quite aptly to HTTP, and in the
>> HTTP context, you can certainly use OAuth if you like.
>>
>
>
> Help me understand this part.
>
> Why can an HTTP request transport a token to a resource server to be able
> to access a specific resource, while SIP request cannot transport a simil=
ar
> token to a SIP server to access a service?
>
>
> Of course both SIP and HTTP can carry arbitrary chunks of data, including
> any sort of token we can imagine. My point is that the semantics of an
> INVITE are really different from the semantics of a POST - and that is wh=
y,
> ultimately, protocols like CCMP don't use SIP as a transport but instead
> use HTTP.  INVITEs are sent by a user agent when it is trying to set up a
> media session between itself and some remote resource on the Internet.
>

Not necessarily; an INVITE is used to create a conference focus, for
example.
And, as Christer mentioned, the use of token is not limited to INVITEs. It
will be used with a REGISTER request to allow a UA to register with the
registrar, and maybe other SIP methods.



> While RFC4579 has a way to open a conference as a side effect of sending
> an INVITE to connect to the conference, the policies associated with the
> sorts of attributes you specified are set totally outside the context of
> that dialog, and the execution of those policies is completely disconnect=
ed
> from the authorization of the conference-opening INVITE. More on this bel=
ow.
>
>  I think the interesting question about standardizing an approach to
> authorization of SIP requests is about INVITE. Of course, SIP supports
> faux-HTTP semantics with mechanisms like PUBLISH and NOTIFY, and we could
> talk about authorizing a number of other non-dialog-forming methods as
> well, but if we're going to kick off an effort to specify authorization
> policy for SIP, the primary thing we need to worry about is INVITE.
>
>
> You even will find some text about Role-Based Access Control in the XCON
>> (RFC6503) security consideration. Although you can use SIP and SDP to se=
t
>> up a BFCP connection (RFC4583), the security of BFCP relies on mutual TL=
S
>> between its own endpoints. It is unclear to me what carrying attribute
>> information around in a SIP INVITE would accomplish with regards to
>> authorizing CCMP or BFCP transactions.
>>
>
> SIP INVITE will carry attributes to authorize that specific SIP request;
> there is nothing in proposal #3 that suggest that SIP will carry
> authorization for CCMP or any other protocol.
>
>
> That is where I think you and I have a disconnect. The SIP INVITE in the
> conference control (RFC4579) case would not carry attributes like the one=
s
> you've identified in order to authorize that specific SIP request. The
> attributes that you identified - the media types the conference allows an=
d
> the number of participants permitted simultaneously in a conference - hav=
e
> nothing to do with the SIP INVITE that opens the conference bridge.
>

Well, not exactly.
The first INVITE that is sent to the conference factory is exactly that.



> Instead, they are components of conference policy. Those policies apply t=
o
> authorization decisions regarding later participants that will be added t=
o
> the conference, and thus to the authorization of later SIP transactions a=
nd
> dialogs, such as when a new participant sends an INVITE to the conference
> bridge after having been sent a REFER by the conference controller, say,
> and the focus decides to reject that INVITE because the maximum number of
> participants has been exceeded.
>

Sure, the INVITE that is sent to the focus would fall under this
description. Why is that a bad thing?



> We can imagine a way to publish such a conference policy to the conferenc=
e
> bridge with SIP, of course, but the question is, in what sense are those
> policies actually aligned with the question of whether or not we should
> authorize the SIP request that carries them? If they are not, then we
> aren't really talking about authorizing SIP requests, we're talking
> uploading long-term application policies over SIP with faux-HTTP semantic=
s,
> which is pretty far from where we started and where I think we need to go=
.
>

There is no "uploading long-term application policies"; the long term
policies are set on the authorization server.
The INVITE will carry the authorization for a specific focus targeted by
the this request.


>
>>
>> So again, the point of my last mail was that the problem of authorizing
>> SIP INVITEs is actually different from the problem of authorizing generi=
c
>> application transactions. SIP INVITEs still have more or less the semant=
ics
>> of a call setup message: they are requesting that a session be initiated=
.
>> Conference control is not managed with SIP transactions, and the securit=
y
>> of conference control protocols does not devolve to authorizing SIP
>> transactions.
>>
>
> Well, RFC4579 defines a SIP Conference Control mechanism that allows you
> to initiate and control a conference server using only SIP requests; so I
> am not sure I understand this statement, unless we have different
> definitions for the term =E2=80=9Cconference control=E2=80=9D.
>
>
> Yeah, I was using the term loosely - once we bring RFC4579 into the mix,
> we need to distinguish "conference control" from "conference policy," and
> then say that RFC4579 and the entire RFC4353 framework is about control,
> not about managing conference policy over SIP. The two attributes you
> propose belong to that conference policy realm rather than conference
> control, strictly speaking.
>
> In any case, my point was not to discuss conference servers specifically,
> but to discuss the general idea of delegating both AuthN and AuthZ in a S=
IP
> network.
>
>
> In that case, if you could furnish a use case with attributes that would
> be carried in a SIP iNVITE that are actually salient to the authorization
> of that particular SIP request, we'd be getting a lot closer to common
> ground.
>
>
An obvious example would be an INVITE with a token sent to a conference
factory to authorize the creation of a focus.

Regards,
 Rifaat



> Jon Peterson
> Neustar, Inc.
>
>
>
> Regards,
>
>  Rifaat
>
>
>
>
>
>> I suspect we'll find that most of the other security attributes that
>> "applications" need to do their jobs similarly don't map onto the semant=
ics
>> of a SIP INVITE - but, I'm still open to entertaining examples that migh=
t
>> cast doubt on that suspicion.
>>
>> Jon Peterson
>> Neustar, Inc.
>>
>> From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
>> Date: Thursday, June 9, 2016 at 10:59 AM
>> To: Jon Peterson <jon.peterson@neustar.biz>
>> Cc: Christer Holmberg <christer.holmberg@ericsson.com>, "sipcore@ietf.or=
g"
>> <sipcore@ietf.org>
>> Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
>>
>> Hi Jon,
>>
>> Sorry for the delayed reply.
>>
>> Let's take the video conference example, and discuss the possible ways
>> for the conference server to provide service.
>>
>>
>>          Trust Domain 1              :    Internet     :  Trust Domain 2
>>                                      :                 :
>>                       +--------+     :                 :
>>                       | IdP    |     :                 :
>>         +-------------| AuthZ  |     :                 :
>>         |             |        |     :                 :
>>         |             +--------+     :                 :
>>         |                 |          :                 :
>>         |                 |          :                 :
>>     +--------+        +--------+     :                 :     +--------+
>>     |        |        | SIP    |     :                 :     | SIP App|
>>     |   UA   |--------| Proxy  |-----:-----------------:-----| Video  |
>>     |        |        |        |     :                 :     | Conferenc=
e
>>     +--------+        +--------+     :                 :     +--------+
>>                                      :                 :
>>
>> Possible solutions:
>>
>> 1. Conference Server acts as an AuthN and AuthZ server. Requests to
>> the server must provide an identity and credentials, and services will
>> be provided based on locally stored information associated with
>> the identity provided in the request.
>>
>> 2. Conference Server acts as an AuthZ server, but delegates the AuthN
>> part. Requests to the server must provide an identity token, and service=
s
>> will be provided based on locally stored information associated with
>> the identity provided in the token.
>>
>> 3. Conference Server delegates both AuthN and AuthZ parts. Requests to
>> the server must provide an access token, which might or might not
>> include information about the user requesting the service, and services
>> will be provided based on information provided in the access token or as
>> a result of an introspection step using the access token.
>>
>>
>> Now, let's look at the 3rd option a bit more, and assume that the
>> server provides the following types of conference services:
>> 1. audio conference
>> 2. audio/video conference
>> 3. web conference
>>
>> or some combination of the above.
>>
>> Let's also assume that the server allows control over the number
>> of participants allowed in each one of these conference types or
>> combinations.
>>
>>
>> Here is one way that I envisioned we could use to address this use case:
>>
>> When an enterprise user in the 1st trust domain wants to register with
>> the SIP Proxy, it will have to authenticate to the IdP and as a result
>> either the UA or the Proxy (depends on the UA's capabilities) will be
>> provided with an AuthN and AuthZ tokens. This process could involve an
>> interim step to allow only the proxy to get access to the tokens, or the
>> tokens could be sent to the UA which will then use them to communicate w=
ith
>> the proxy.
>>
>> The AuthN token is the identity of the user, and the AuthZ token has
>> all the services granted to the user by the IdP/AuthZ Service.
>>
>> When the enterprise user needs access to the conference service, the
>> UA will send an INVITE to the Proxy, and will include the AuthZ token if=
 it
>> is in possession of the tokens; otherwise, the proxy will add the
>> appropriate token to the outgoing INVITE sent to the conference server.
>>
>> The scope of the AuthZ token might grant the user access to
>> multiple services; to avoid sending that AuthZ token to all servers
>> providing services granted to the user, the proxy could obtain a differe=
nt
>> token with limited scope that is specific to the service being provided
>> (conference services in this case); there is a standard Token Exchange
>> mechanism for that.
>>
>> The AuthZ token will include information about the type of service
>> allowed (audio, audio/video, etc), and the limit on the number of
>> participants. The AuthZ token might or might not have any information ab=
out
>> the user requesting the service, which depends on the level of privacy
>> needed.
>>
>> When the conference server receives this token in the SIP INVITE, all
>> it has to do is validate that the token was signed by an IdP that it has
>> a trust relationship with, and provide the service requested by the
>> AuthZ token. The conference server does not need the AuthN token to be a=
ble
>> to provide a service.
>>
>> Regards,
>>  Rifaat
>>
>>
>> On Wed, May 25, 2016 at 3:06 PM, Peterson, Jon <jon.peterson@neustar.biz=
>
>> wrote:
>>
>>>
>>> Also, if we decided to separate the AuthN and AuthZ, then we might need
>>> to carry more than one token.
>>>
>>>
>>> I could see an argument that there is a need to carry the identity of
>>> the user in one header, and a reference to a place where you can get
>>> attributes about the originating user in another header, say. That's mo=
re
>>> or less what we envisioned the last time we went down this path, which =
led
>>> to the RFC4484 requirements, largely informed by SAML (which I seem to
>>> recall 3GPP was interested in at the time).
>>>
>>> But it's no accident that the work on RFC4484 never went beyond
>>> requirements. Today, I think that much of the useful work that could be
>>> done on the authorization question is in identifying which SIP
>>> authorization policies are likely to depend on attributes as opposed to
>>> just the identity of the end user. While we can readily conceive of
>>> applications adjunct to communications that would require attributes, t=
he
>>> challenge has been justifying why you would invoke those applications w=
ith
>>> SIP rather than, say, HTTP.  The fundamental semantics of a SIP INVITE =
have
>>> remained pretty close to those of placing a telephone call.  The
>>> authorization policies of a telephone call are really different than th=
at
>>> of a web service, say, both in terms of how and why they are invoked, w=
hat
>>> kinds of parties are involved, what forms redirection or cross-domain
>>> interaction can occur. With appropriate caveats for other methods (nota=
bly
>>> MESSAGE, SUBSCRIBE/NOTIFY. INFO), SIP requests have authorization polic=
y
>>> requirements that look a lot like call treatment policies.
>>>
>>> That much said, I am kind of interested in call treatment policies, and
>>> their interaction with identity information. But just saying that there
>>> might be a video conference service in the cloud, say, doesn't shed a l=
ot
>>> of light for me. If there's a video conference, I'd ordinarily think yo=
u
>>> just call into it. If the user who opens the conference bridge needs to=
 be
>>> charged, that's identity information - which in its usual form will con=
vey
>>> the individual user and their domain, like user@domain. If you need
>>> more complex semantics than that, it's probably because you're doing
>>> something more complicated than just calling in to the conference. But =
it's
>>> less clear to me why you'd use SIP for such cases. Drilling down deeper
>>> into these use cases would help me understand where we're going, anyway=
 -
>>> the use cases in RFC4484 Section 4, which largely involve billing
>>> (including when dealing with various protocol gateways) and resource
>>> management (both preemption and priority), ultimately didn't seem
>>> sufficient to motivate protocol work.
>>>
>>> Another issues in the need, in some cases, to avoid sending the token t=
o
>>> the UA; in this case there is a need for an interim step to get the
>>> token(s) to the SIP proxy.
>>>
>>>
>>> If this information is carried in headers, then presumably it could be
>>> added, consumed, subtracted, or whatever by any relevant intermediaries=
.
>>>
>>> Jon Peterson
>>> Neustar, Inc.
>>>
>>
>>
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Tue, Jun 14, 2016 at 1:27 PM, Peterson, Jon <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:jon.peterson@neustar.biz" target=3D"_blank">jon.peterson@ne=
ustar.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-style:solid;=
border-left-color:rgb(204,204,204);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>This time my response should be more complete. Sorry about that!</div>=
<span class=3D"">
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div>
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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>Radhika sent a helpful summary of conference control protocols to the =
list which surveyed mechanisms like CCMP, BFCP, and MRB, protocols which ac=
tually let you manage attributes of a conference like the number of partici=
pants and media types allowed.
</div>
</div>
</blockquote>
<div><br>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">This conference use case was an e=
xample for a SIP-based conference server (RFC4579), just to describe the id=
ea around the delegation of AuthNZ.</font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">If the server is a CCMP-based con=
ference, then you would obviously use that to control and manage the confer=
ence server.</font></p>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span><div>
<div>While RFC4579 (an implementation of the RFC4353 framework) provides a =
means to do conference control from within SIP, the kinds of control primit=
ives it specifies are totally decoupled from the sorts of policies that the=
 attributes you proposed would constrain.
 Those policies fall under what RFC4353 calls &quot;conference policy.&quot=
; </div></div></div></blockquote><div><br></div><div>I agree. This is an im=
portant distinction and clarification.</div><div><br></div><div>=C2=A0</div=
><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border=
-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);=
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>Neither RFC4579 nor its=
 requirements (RFC4245) really say anything about how conference policy is =
set or distributed. RFC4353 does, though, and what it says is (in 4.6) is &=
quot;the conference policy
 can be manipulated using web applications or voice applications [i.e. IVRs=
 - JFP]. It can also be manipulated with non-SIP-specific standard or propr=
ietary protocols.&quot;</div>
<div><br></div></div></div></blockquote><div><br></div><div>Proposal #3 del=
egates this function from the conference server to the authorization server=
. The conference server becomes a consumer and enforcer of a policy.</div><=
div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"m=
argin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;borde=
r-left-color:rgb(204,204,204);padding-left:1ex"><div style=3D"word-wrap:bre=
ak-word;color:rgb(0,0,0);font-size:14px;font-family:Calibri,sans-serif"><di=
v><div>
</div>
<div>Authorizing SIP requests is misaligned with setting conference policy,=
 especially in consideration of the sorts of attributes you seem to have in=
 mind - even when we&#39;re considering controlling conferences compliant w=
ith the RFC4353 framework.=C2=A0 That&#39;s because
 the conference in RFC4579 isn&#39;t &quot;created&quot; when a user agent =
sends an INVITE request to the conference bridge - it is created when the c=
onference URI is created, which is necessarily beforehand (or at best, ad h=
oc through the conference factory). Conference
 policy is set and disseminated in that context, not in the context of the =
SIP transaction that happens to open the bridge.</div></div></div></blockqu=
ote><div><br></div><div>In the diagram I provided, I did not separate the C=
onference Factory from the Focus, maybe I should have.</div><div>The INVITE=
 with the token will be sent to the factory, which will create the focus an=
d set the proper limitation on the focus based on the authorization provide=
d in the token.</div><div>How is this misaligned with settings a conference=
 policy?</div><div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_qu=
ote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-st=
yle:solid;border-left-color:rgb(204,204,204);padding-left:1ex"><div style=
=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-family:Calibr=
i,sans-serif"><div>
<div><br>
</div>
<div>Moreover, my point was that the example of CCMP shows that you can do =
conference control with OAuth over in the HTTP realm without needing to add=
 anything whatsoever to SIP.=C2=A0 If you need OAuth-style controls for con=
ferencing, that would seem to be a quicker
 path to implementation.</div></div></div></blockquote><div><br></div><div>=
If your conference server supports CCMP, then yes, that would be the right =
answer. What if your server does not support CCMP?<br></div><div>But, again=
, the conference use case is an example only to drive the point, not to dis=
cuss the best way to implement conference server.</div><div><br></div><div>=
=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0=
.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb(20=
4,204,204);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><span class=3D"">
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
=C2=A0</p>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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>The interesting thing about protocols like CCMP is that they are not, =
you know, actually transported by SIP. CCMP is transported by HTTP. Why? Be=
cause the semantics of conference control doesn&#39;t actually map onto SIP=
 INVITE transactions very well. They
 map quite aptly to HTTP, and in the HTTP context, you can certainly use OA=
uth if you like.</div>
</div>
</blockquote>
<div>=C2=A0</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">Help me understand this part.</fo=
nt></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">Why can an HTTP request transport=
 a token to a resource server to be able to access a specific resource, whi=
le SIP request cannot transport a similar token to a SIP server to access a=
 service?</font></p>
</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span><div>
<div>Of course both SIP and HTTP can carry arbitrary chunks of data, includ=
ing any sort of token we can imagine. My point is that the semantics of an =
INVITE are really different from the semantics of a POST - and that is why,=
 ultimately, protocols like CCMP
 don&#39;t use SIP as a transport but instead use HTTP.=C2=A0 INVITEs are s=
ent by a user agent when it is trying to set up a media session between its=
elf and some remote resource on the Internet. </div></div></div></blockquot=
e><div><br></div><div>Not necessarily; an INVITE is used to create a confer=
ence focus, for example.</div><div>And, as Christer mentioned, the use of t=
oken is not limited to INVITEs. It will be used with a REGISTER request to =
allow a UA to register with the registrar, and maybe other SIP methods.</di=
v><div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=
=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;=
border-left-color:rgb(204,204,204);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><div>While RFC4579 has a way to open a conference as a side effect o=
f sending
 an INVITE to connect to the conference, the policies associated with the s=
orts of attributes you specified are set totally outside the context of tha=
t dialog, and the execution of those policies is completely disconnected fr=
om the authorization of the conference-opening
 INVITE. More on this below.</div>
<div><br>
</div>
<div>=C2=A0I think the interesting question about standardizing an approach=
 to authorization of SIP requests is about INVITE. Of course, SIP supports =
faux-HTTP semantics with mechanisms like PUBLISH and NOTIFY, and we could t=
alk about authorizing a number of other
 non-dialog-forming methods as well, but if we&#39;re going to kick off an =
effort to specify authorization policy for SIP, the primary thing we need t=
o worry about is INVITE.</div>
</div><span class=3D"">
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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>You even will find some text about Role-Based Access Control in the XC=
ON (RFC6503) security consideration. Although you can use SIP and SDP to se=
t up a BFCP connection (RFC4583), the security of BFCP relies on mutual TLS=
 between its own endpoints. It is
 unclear to me what carrying attribute information around in a SIP INVITE w=
ould accomplish with regards to authorizing CCMP or BFCP transactions.</div=
>
</div>
</blockquote>
<div><br>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">SIP INVITE will carry attributes =
to authorize that specific SIP request; there is nothing in proposal #3 tha=
t suggest that SIP will carry authorization for CCMP or any other protocol.=
</font></p>
</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span><div>
<div>That is where I think you and I have a disconnect. The SIP INVITE in t=
he conference control (RFC4579) case would not carry attributes like the on=
es you&#39;ve identified in order to authorize that specific SIP request. T=
he attributes that you identified -
 the media types the conference allows and the number of participants permi=
tted simultaneously in a conference - have nothing to do with the SIP INVIT=
E that opens the conference bridge. </div></div></div></blockquote><div><br=
></div><div>Well, not exactly.</div><div>The first INVITE that is sent to t=
he conference factory is exactly that.</div><div><br></div><div>=C2=A0<br><=
/div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;bo=
rder-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,2=
04);padding-left:1ex"><div style=3D"word-wrap:break-word;color:rgb(0,0,0);f=
ont-size:14px;font-family:Calibri,sans-serif"><div><div>Instead, they are c=
omponents of conference policy. Those policies apply
 to authorization decisions regarding later participants that will be added=
 to the conference, and thus to the authorization of later SIP transactions=
 and dialogs, such as when a new participant sends an INVITE to the confere=
nce bridge after having been sent
 a REFER by the conference controller, say, and the focus decides to reject=
 that INVITE because the maximum number of participants has been exceeded.<=
/div></div></div></blockquote><div><br></div><div>Sure, the INVITE that is =
sent to the focus would fall under this description. Why is that a bad thin=
g?</div><div>=C2=A0</div><div><br></div><blockquote class=3D"gmail_quote" s=
tyle=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:so=
lid;border-left-color:rgb(204,204,204);padding-left:1ex"><div style=3D"word=
-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-family:Calibri,sans-s=
erif"><div>
<div><br>
</div>
<div>We can imagine a way to publish such a conference policy to the confer=
ence bridge with SIP, of course, but the question is, in what sense are tho=
se policies actually aligned with the question of whether or not we should =
authorize the SIP request that carries
 them? If they are not, then we aren&#39;t really talking about authorizing=
 SIP requests, we&#39;re talking uploading long-term application policies o=
ver SIP with faux-HTTP semantics, which is pretty far from where we started=
 and where I think we need to go.</div></div></div></blockquote><div><br></=
div><div>There is no &quot;<span style=3D"color:rgb(0,0,0);font-family:Cali=
bri,sans-serif;font-size:14px">uploading long-term application policies</sp=
an>&quot;; the long term policies are set on the authorization server.=C2=
=A0</div><div>The INVITE will carry the authorization for a specific focus =
targeted by the this request.</div><div><br></div><blockquote class=3D"gmai=
l_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-lef=
t-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex"><div sty=
le=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-family:Cali=
bri,sans-serif"><div>
</div><span class=3D"">
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<span style=3D"font-family:Arial,sans-serif;font-size:12pt"></span></p>
</div>
<div>=C2=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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>So again, the point of my last mail was that the problem of authorizin=
g SIP INVITEs is actually different from the problem of authorizing generic=
 application transactions. SIP INVITEs still have more or less the semantic=
s of a call setup message: they
 are requesting that a session be initiated. Conference control is not mana=
ged with SIP transactions, and the security of conference control protocols=
 does not devolve to authorizing SIP transactions.
</div>
</div>
</blockquote>
<div><br>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">Well, RFC4579 defines a SIP Confe=
rence Control mechanism that allows you to initiate and control a conferenc=
e server using only SIP requests; so I am not sure I understand this statem=
ent, unless we have different definitions
 for the term =E2=80=9Cconference control=E2=80=9D.</font></p>
</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span><div>Yeah, I was using the term loosely - once we bring RFC4579 into=
 the mix, we need to distinguish &quot;conference control&quot; from &quot;=
conference policy,&quot; and then say that RFC4579 and the entire RFC4353 f=
ramework is about control, not about managing conference policy
 over SIP. The two attributes you propose belong to that conference policy =
realm rather than conference control, strictly speaking.</div><span class=
=3D"">
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<span style=3D"font-family:arial,helvetica,sans-serif">In any case, my poin=
t was not to discuss conference servers specifically, but to discuss the ge=
neral idea of delegating both AuthN and AuthZ in a SIP network.</span></p>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span><div>
<div>In that case, if you could furnish a use case with attributes that wou=
ld be carried in a SIP iNVITE that are actually salient to the authorizatio=
n of that particular SIP request, we&#39;d be getting a lot closer to commo=
n ground.</div>
<div><br></div></div></div></blockquote><div><br></div><div>An obvious exam=
ple would be an INVITE with a token sent to a conference factory to authori=
ze the creation of a focus.</div><div><br></div><div>Regards,</div><div>=C2=
=A0Rifaat</div><div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_q=
uote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-s=
tyle:solid;border-left-color:rgb(204,204,204);padding-left:1ex"><div style=
=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-family:Calibr=
i,sans-serif"><div><div>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">=C2=A0</font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">Regards,</font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<font face=3D"arial,helvetica,sans-serif">=C2=A0Rifaat</font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<span style=3D"font-size:12pt;font-family:Arial,sans-serif"><br>
</span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial">
<span style=3D"font-size:12pt;font-family:Arial,sans-serif"><br>
</span></p>
</div><div><div class=3D"h5">
<div>=C2=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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 suspect we&#39;ll find that most of the other security attributes th=
at &quot;applications&quot; need to do their jobs similarly don&#39;t map o=
nto the semantics of a SIP INVITE - but, I&#39;m still open to entertaining=
 examples that might cast doubt on that suspicion.=C2=A0</div>
<div><br>
</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, June 9, 2016 at 10:=
59 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>Christer Holmberg &lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmb=
erg@ericsson.com</a>&gt;, &quot;<a href=3D"mailto:sipcore@ietf.org" target=
=3D"_blank">sipcore@ietf.org</a>&quot; &lt;<a href=3D"mailto:sipcore@ietf.o=
rg" target=3D"_blank">sipcore@ietf.org</a>&gt;<span><br>
<span style=3D"font-weight:bold">Subject: </span>Re: [sipcore] SIP AuthNZ P=
roblem Statement - v3<br>
</span></div>
<div>
<div>
<div><br>
</div>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div>
<div>
<div dir=3D"ltr"><font face=3D"monospace,monospace">Hi Jon,</font>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Sorry for the delayed reply.</font>=
</div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div>
<div>
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><font face=3D"monospace,monospace">Let&#39;s take the video conference=
 example, and discuss the possible ways for=C2=A0</font><span style=3D"font=
-family:monospace,monospace">the conference server to provide service.</spa=
n></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0T=
rust Domain 1 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0: =C2=A0 =C2=
=A0Internet =C2=A0 =C2=A0 : =C2=A0Trust Domain 2</font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 :</font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 +--------+ =C2=A0 =C2=A0 : =C2=A0=
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :</font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | IdP =C2=A0 =C2=A0| =C2=A0 =C2=
=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :</font></div=
>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 +------=
-------| AuthZ =C2=A0| =C2=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 :</font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0 =C2=A0| =C2=
=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :</fon=
t></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 +--------+ =C2=A0 =C2=A0 : =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :</font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0=
 =C2=A0 =C2=A0: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :</=
font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0=
 =C2=A0 =C2=A0: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :</=
font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 +--------+ =C2=A0 =C2=
=A0 =C2=A0 =C2=A0+--------+ =C2=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 : =C2=A0 =C2=A0 +--------+</font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=
=A0 =C2=A0| =C2=A0 =C2=A0 =C2=A0 =C2=A0| SIP =C2=A0 =C2=A0| =C2=A0 =C2=A0 :=
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : =C2=A0 =C2=A0 | =
SIP App|</font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 | =C2=A0 UA =C2=A0 |-=
-------| Proxy =C2=A0|-----:-----------------:-----| Video =C2=A0|</font></=
div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=
=A0 =C2=A0| =C2=A0 =C2=A0 =C2=A0 =C2=A0| =C2=A0 =C2=A0 =C2=A0 =C2=A0| =C2=
=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : =C2=
=A0 =C2=A0 | Conference</font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 +--------+ =C2=A0 =C2=
=A0 =C2=A0 =C2=A0+--------+ =C2=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 : =C2=A0 =C2=A0 +--------+</font></div>
<div><font face=3D"monospace,monospace">=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 :</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Possible solutions:</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">1. Conference Server acts as an Aut=
hN and AuthZ server. Requests to the=C2=A0server must provide an identity a=
nd credentials, and services will be=C2=A0provided based on locally stored =
information associated with the=C2=A0identity provided
 in the request.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">2. Conference Server acts as an Aut=
hZ server, but delegates the AuthN part.=C2=A0Requests to the server must p=
rovide an identity token, and services will=C2=A0be provided based on local=
ly stored information associated with the=C2=A0identity
 provided in the token.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">3. Conference Server delegates both=
 AuthN and AuthZ parts. Requests to the=C2=A0server must provide an access =
token, which might or might not include=C2=A0information about the user req=
uesting the service, and services will=C2=A0be provided
 based on information provided in the access token or as a=C2=A0result of a=
n introspection step using the access token.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Now, let&#39;s look at the 3rd opti=
on a bit more, and assume that the server=C2=A0provides the following types=
 of conference services:</font></div>
<div><font face=3D"monospace,monospace">1. audio conference</font></div>
<div><font face=3D"monospace,monospace">2. audio/video conference</font></d=
iv>
<div><font face=3D"monospace,monospace">3. web conference</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">or some=C2=A0combination=C2=A0of th=
e above.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Let&#39;s also assume that the serv=
er allows control over the number of=C2=A0participants allowed in each one =
of these conference types or combinations.</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Here is one way that I envisioned w=
e could use to address this use case:</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
</div>
</div>
</div>
<font face=3D"monospace,monospace">When an enterprise user in the 1st trust=
 domain wants to register with the=C2=A0SIP Proxy, it will have to authenti=
cate to the IdP and as a result either=C2=A0the UA or the Proxy (depends on=
 the UA&#39;s capabilities) will be provided=C2=A0with
 an AuthN and AuthZ tokens. This process could involve an interim step=C2=
=A0to allow only the proxy to get access to the tokens, or the tokens could=
 be=C2=A0sent to the UA which will then use them to communicate with the pr=
oxy.</font>
<div><font face=3D"monospace,monospace"><br>
The AuthN token is the identity of the user, and the AuthZ token has all=C2=
=A0the services granted to the user by the IdP/AuthZ Service.<br>
<br>
When the enterprise user needs access to the conference service, the UA=C2=
=A0will send an INVITE to the Proxy, and will include the AuthZ token if it=
 is=C2=A0in possession of the tokens; otherwise, the proxy will add the app=
ropriate=C2=A0token to the outgoing INVITE sent
 to the conference server.<br>
<br>
The scope of the AuthZ token might grant the user access to multiple=C2=A0s=
ervices; to avoid sending that AuthZ token to all servers providing=C2=A0se=
rvices granted to the user, the proxy could obtain a different token with=
=C2=A0limited scope that is specific to the service
 being provided (conference=C2=A0services in this case); there is a standar=
d Token Exchange mechanism for=C2=A0that.<br>
<br>
The AuthZ token will include information about the type of service allowed=
=C2=A0(audio, audio/video, etc), and the limit on the number of participant=
s.=C2=A0The AuthZ token might or might not have any information about the u=
ser=C2=A0requesting the service, which depends on
 the level of privacy needed.<br>
<br>
When the conference server receives this token in the SIP INVITE, all it=C2=
=A0has to do is validate that the token was signed by an IdP that it has a=
=C2=A0trust relationship with, and provide the service requested by the Aut=
hZ=C2=A0token. The conference server does not need
 the AuthN token to be able to=C2=A0provide a service.</font>
<div>
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><font face=3D"monospace,monospace"><br>
</font></div>
<div><font face=3D"monospace,monospace">Regards,</font></div>
<div><font face=3D"monospace,monospace">=C2=A0Rifaat</font></div>
<div><font face=3D"monospace,monospace"><br>
</font></div>
</div>
</div>
</div>
</div>
</div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Wed, May 25, 2016 at 3:06 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-style:solid;border-left-color:rgb(204,204,204);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">
<span><span>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
</div>
</div>
</div>
</span><span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>Also, if we decided to separate the AuthN and AuthZ, then we might nee=
d to carry more than one token.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span>
<div>I could see an argument that there is a need to carry the identity of =
the user in one header, and a reference to a place where you can get attrib=
utes about the originating user in another header, say. That&#39;s more or =
less what we envisioned the last time
 we went down this path, which led to the RFC4484 requirements, largely inf=
ormed by SAML (which I seem to recall 3GPP was interested in at the time).<=
/div>
<div><br>
</div>
<div>But it&#39;s no accident that the work on RFC4484 never went beyond re=
quirements. Today, I think that much of the useful work that could be done =
on the authorization question is in identifying which SIP authorization pol=
icies are likely to depend on attributes
 as opposed to just the identity of the end user. While we can readily conc=
eive of applications adjunct to communications that would require attribute=
s, the challenge has been justifying why you would invoke those application=
s with SIP rather than, say, HTTP.
 =C2=A0The fundamental semantics of a SIP INVITE have remained pretty close=
 to those of placing a telephone call.=C2=A0 The authorization policies of =
a telephone call are really different than that of a web service, say, both=
 in terms of how and why they are invoked,
 what kinds of parties are involved, what forms redirection or cross-domain=
 interaction can occur. With appropriate caveats for other methods (notably=
 MESSAGE, SUBSCRIBE/NOTIFY. INFO), SIP requests have authorization policy r=
equirements that look a lot like
 call treatment policies.</div>
<div><br>
</div>
<div>That much said, I am kind of interested in call treatment policies, an=
d their interaction with identity information. But just saying that there m=
ight be a video conference service in the cloud, say, doesn&#39;t shed a lo=
t of light for me. If there&#39;s a video
 conference, I&#39;d ordinarily think you just call into it. If the user wh=
o opens the conference bridge needs to be charged, that&#39;s identity info=
rmation - which in its usual form will convey the individual user and their=
 domain, like user@domain. If you need more
 complex semantics than that, it&#39;s probably because you&#39;re doing so=
mething more complicated than just calling in to the conference. But it&#39=
;s less clear to me why you&#39;d use SIP for such cases. Drilling down dee=
per into these use cases would help me understand
 where we&#39;re going, anyway - the use cases in RFC4484 Section 4, which =
largely involve billing (including when dealing with various protocol gatew=
ays) and resource management (both preemption and priority), ultimately did=
n&#39;t seem sufficient to motivate protocol
 work.</div>
<span>
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>Another issues in the need, in some cases, to avoid sending the token =
to the UA; in this case there is a need for an interim step to get the toke=
n(s) to the SIP proxy.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span>
<div>If this information is carried in headers, then presumably it could be=
 added, consumed, subtracted, or whatever by any relevant intermediaries.=
=C2=A0</div>
<span><font color=3D"#888888">
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
</font></span></div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</blockquote>
</div>
</div>
</span></div>
</blockquote>
</div></div></div>
<br>
</div>
</div>
</blockquote>
</span>
</div>

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

--001a1143e6e414dcbd0535524206--


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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Thread-Topic: [sipcore] SIP AuthNZ Problem Statement - v3
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Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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--_000_D386CD3B19C95Fjonpetersonneustarbiz_
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In that case, if you could furnish a use case with attributes that would be=
 carried in a SIP iNVITE that are actually salient to the authorization of =
that particular SIP request, we'd be getting a lot closer to common ground.


An obvious example would be an INVITE with a token sent to a conference fac=
tory to authorize the creation of a focus.

We're still talking past each other. My point above was that the two attrib=
utes you specified - constraints on conference size and media types - have =
nothing to do with the authorization of the INVITE carrying those attribute=
s. Like, unless the conference size permitted is 0, then the INVITE that cr=
eates an ad hoc conference and connects you to its bridge is not going to b=
e rejected by the focus on the basis of that conference size attribute. If =
the maximum conference size were 0, the AuthZ service just wouldn't attach =
a token to an INVITE as it would never be allowed to start a conference. Sa=
me goes for media types.

Instead, what your two attributes constrain is what happens after the confe=
rence is created: when ten users try to dial in to a conference with a perm=
itted size of nine, say. That tenth INVITE would get rejected - not the con=
ference-forming INVITE that you propose would carry the token. That's why w=
hat you're proposing is overloading the INVITE to tunnel conference policy =
constraints to the focus. I mean, maybe that's a problem space worth invest=
igating, though I suspect there are good reasons why RFC4353 specified that=
 conference policy is not manipulated with SIP - mostly because the semanti=
cs of manipulating conference policy probably don't map well onto dialog-fo=
rming SIP requests. But I think that conference policy provisioning problem=
 space would be pretty far from where we started in this discussion.

Let me reiterate what I wrote in my previous mail, which I quote here at th=
e top: "furnish a use case with attributes that would be carried in a SIP i=
NVITE that are actually salient to the authorization of that particular SIP=
 request." Again: "that particular SIP request." Not later transactions in =
some broader context in which the original user agent probably isn't even a=
n endpoint.

The Authorization header in SIP, for example, is actually used by the UAS t=
o make a decision about whether or not to authorize the request in which th=
e header appears.  The token that you propose a SIP INVITE would carry in t=
his case is not used by the relying party to make a decision about whether =
or not that INVITE is authorized. That's why I believe this conference poli=
cy issue is a different problem space entirely from figuring out how to imp=
lement attribute-based authorization for SIP requests.

Jon Peterson
Neustar, Inc.


Regards,
 Rifaat


--_000_D386CD3B19C95Fjonpetersonneustarbiz_
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<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;">
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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">
<span class=3D""><br>
<br>
</span>
<div>
<div>In that case, if you could furnish a use case with attributes that wou=
ld be carried in a SIP iNVITE that are actually salient to the authorizatio=
n of that particular SIP request, we'd be getting a lot closer to common gr=
ound.</div>
<div><br>
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>An obvious example would be an INVITE with a token sent to a conferenc=
e factory to authorize the creation of a focus.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>We're still talking past each other. My point above was that the two a=
ttributes you specified - constraints on conference size and media types - =
have nothing to do with the authorization of the INVITE carrying those attr=
ibutes. Like, unless the conference
 size permitted is 0, then the INVITE that creates an ad hoc conference and=
 connects you to its bridge is not going to be rejected by the focus on the=
 basis of that conference size attribute. If the maximum conference size we=
re 0, the AuthZ service just wouldn't
 attach a token to an INVITE as it would never be allowed to start a confer=
ence. Same goes for media types.</div>
<div><br>
</div>
<div>Instead, what your two attributes constrain is what happens after the =
conference is created: when ten users try to dial in to a conference with a=
 permitted size of nine, say. That tenth INVITE would get rejected - not th=
e conference-forming INVITE that
 you propose would carry the token. That's why what you're proposing is ove=
rloading the INVITE to tunnel conference policy constraints to the focus. I=
 mean, maybe that's a problem space worth investigating, though I suspect t=
here are good reasons why RFC4353
 specified that conference policy is not manipulated with SIP - mostly beca=
use the semantics of manipulating conference policy probably don't map well=
 onto dialog-forming SIP requests. But I think that conference policy provi=
sioning problem space would be pretty
 far from where we started in this discussion.</div>
<div><br>
</div>
<div>Let me reiterate what I wrote in my previous mail, which I quote here =
at the top: &quot;furnish a use case with attributes that would be carried =
in a SIP iNVITE that are actually salient to the authorization of that part=
icular SIP request.&quot; Again: &quot;that particular
 SIP request.&quot; Not later transactions in some broader context in which=
 the original user agent probably isn't even an endpoint.</div>
<div><br>
</div>
<div>The Authorization header in SIP, for example, is actually used by the =
UAS to make a decision about whether or not to authorize the request in whi=
ch the header appears. &nbsp;The token that you propose a SIP INVITE would =
carry in this case is not used by the
 relying party to make a decision about whether or not that INVITE is autho=
rized. That's why I believe this conference policy issue is a different pro=
blem space entirely from figuring out how to implement attribute-based auth=
orization for SIP requests.</div>
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<div>Regards,</div>
<div>&nbsp;Rifaat</div>
<div><br>
</div>
</div>
</div>
</div>
</blockquote>
</span>
</body>
</html>

--_000_D386CD3B19C95Fjonpetersonneustarbiz_--


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From: worley@ariadne.com (Dale R. Worley)
To: Robert Sparks <rjsparks@nostrum.com>
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This is a great step for cleaning up this mess!

If I'm not mistaken, RFC 3325 (P-Asserted-Identity and
P-Preferred-Identity) has a similar problem.  The grammar is

      PAssertedID = "P-Asserted-Identity" HCOLON PAssertedID-value
                      *(COMMA PAssertedID-value)
      PAssertedID-value = name-addr / addr-spec
      PPreferredID = "P-Preferred-Identity" HCOLON PPreferredID-value
                        *(COMMA PPreferredID-value)
      PPreferredID-value = name-addr / addr-spec

Since there are no header-field-parameters, semicolon is not
problematic, but comma is a problem, as is question mark.

Dale


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From: Brett Tate <brett@broadsoft.com>
References: <e84dc0df-d77c-8521-d797-c6b0393f6efd@nostrum.com> (rjsparks@nostrum.com) <877fdpuc96.fsf@hobgoblin.ariadne.com>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/sipcore/gee8ML2-GuI0UZlNYzZPQar320E>
Subject: Re: [sipcore] draft-sparks-sipcore-name-addr-guidance-00
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Hi,

Yes; it is discussed within RFC 3325 errata (3744 and 3894).

Should it be mentioned within the draft?


> -----Original Message-----
> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Dale R.
Worley
> Sent: Wednesday, June 15, 2016 9:57 PM
> To: Robert Sparks
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] draft-sparks-sipcore-name-addr-guidance-00
>
> This is a great step for cleaning up this mess!
>
> If I'm not mistaken, RFC 3325 (P-Asserted-Identity and
> P-Preferred-Identity) has a similar problem.  The grammar is
>
>       PAssertedID = "P-Asserted-Identity" HCOLON PAssertedID-value
>                       *(COMMA PAssertedID-value)
>       PAssertedID-value = name-addr / addr-spec
>       PPreferredID = "P-Preferred-Identity" HCOLON PPreferredID-value
>                         *(COMMA PPreferredID-value)
>       PPreferredID-value = name-addr / addr-spec
>
> Since there are no header-field-parameters, semicolon is not
problematic, but
> comma is a problem, as is question mark.
>
> Dale
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From nobody Thu Jun 16 04:45:22 2016
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From: "Drage, Keith (Nokia - GB)" <keith.drage@nokia.com>
To: Brett Tate <brett@broadsoft.com>, "Dale R. Worley" <worley@ariadne.com>, Robert Sparks <rjsparks@nostrum.com>, "sipcore@ietf.org" <sipcore@ietf.org>
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Subject: Re: [sipcore] draft-sparks-sipcore-name-addr-guidance-00
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A reminder that RFC 3325 is an informational document, so I am not sure whe=
ther a standards track document can update an informational document. Anyon=
e able to clarify?

Keith

-----Original Message-----
From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Brett Tate
Sent: 16 June 2016 12:03
To: Dale R. Worley; Robert Sparks; sipcore@ietf.org
Subject: Re: [sipcore] draft-sparks-sipcore-name-addr-guidance-00

Hi,

Yes; it is discussed within RFC 3325 errata (3744 and 3894).

Should it be mentioned within the draft?


> -----Original Message-----
> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Dale R.
Worley
> Sent: Wednesday, June 15, 2016 9:57 PM
> To: Robert Sparks
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] draft-sparks-sipcore-name-addr-guidance-00
>
> This is a great step for cleaning up this mess!
>
> If I'm not mistaken, RFC 3325 (P-Asserted-Identity and
> P-Preferred-Identity) has a similar problem.  The grammar is
>
>       PAssertedID =3D "P-Asserted-Identity" HCOLON PAssertedID-value
>                       *(COMMA PAssertedID-value)
>       PAssertedID-value =3D name-addr / addr-spec
>       PPreferredID =3D "P-Preferred-Identity" HCOLON PPreferredID-value
>                         *(COMMA PPreferredID-value)
>       PPreferredID-value =3D name-addr / addr-spec
>
> Since there are no header-field-parameters, semicolon is not
problematic, but
> comma is a problem, as is question mark.
>
> Dale
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

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


From nobody Thu Jun 16 06:19:22 2016
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From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Date: Thu, 16 Jun 2016 09:19:16 -0400
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To: "Peterson, Jon" <jon.peterson@neustar.biz>
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Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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--001a114413c0d1c3bf0535651593
Content-Type: text/plain; charset=UTF-8

The attributes I provided are examples of attributes that might be provided
in an access token; it was not meant to include all the attributes for this
use case, because my intention was not to focus on conference server, but
to discuss the idea in general.

The following could be used with the conference factory to request a
conference focus:
The INVITE sent to the conference factory could include a token that has a
*conference-type* attribute which could take one of the following
values: *audio,
video, webconference, audio-webconference, or video-webconference*.
The conference factory will then issue a *conference uri* for the requested
conference type, assuming the token is valid. If the *conference-type*
attribute is missing, then the request would be rejected.

Is that what you are looking for?

Regards,
 Rifaat





On Wed, Jun 15, 2016 at 5:06 PM, Peterson, Jon <jon.peterson@neustar.biz>
wrote:

>
>>
>> In that case, if you could furnish a use case with attributes that would
>> be carried in a SIP iNVITE that are actually salient to the authorization
>> of that particular SIP request, we'd be getting a lot closer to common
>> ground.
>>
>>
> An obvious example would be an INVITE with a token sent to a conference
> factory to authorize the creation of a focus.
>
>
> We're still talking past each other. My point above was that the two
> attributes you specified - constraints on conference size and media types -
> have nothing to do with the authorization of the INVITE carrying those
> attributes. Like, unless the conference size permitted is 0, then the
> INVITE that creates an ad hoc conference and connects you to its bridge is
> not going to be rejected by the focus on the basis of that conference size
> attribute. If the maximum conference size were 0, the AuthZ service just
> wouldn't attach a token to an INVITE as it would never be allowed to start
> a conference. Same goes for media types.
>
> Instead, what your two attributes constrain is what happens after the
> conference is created: when ten users try to dial in to a conference with a
> permitted size of nine, say. That tenth INVITE would get rejected - not the
> conference-forming INVITE that you propose would carry the token. That's
> why what you're proposing is overloading the INVITE to tunnel conference
> policy constraints to the focus. I mean, maybe that's a problem space worth
> investigating, though I suspect there are good reasons why RFC4353
> specified that conference policy is not manipulated with SIP - mostly
> because the semantics of manipulating conference policy probably don't map
> well onto dialog-forming SIP requests. But I think that conference policy
> provisioning problem space would be pretty far from where we started in
> this discussion.
>
> Let me reiterate what I wrote in my previous mail, which I quote here at
> the top: "furnish a use case with attributes that would be carried in a SIP
> iNVITE that are actually salient to the authorization of that particular
> SIP request." Again: "that particular SIP request." Not later transactions
> in some broader context in which the original user agent probably isn't
> even an endpoint.
>
> The Authorization header in SIP, for example, is actually used by the UAS
> to make a decision about whether or not to authorize the request in which
> the header appears.  The token that you propose a SIP INVITE would carry in
> this case is not used by the relying party to make a decision about whether
> or not that INVITE is authorized. That's why I believe this conference
> policy issue is a different problem space entirely from figuring out how to
> implement attribute-based authorization for SIP requests.
>
> Jon Peterson
> Neustar, Inc.
>
>
> Regards,
>  Rifaat
>
>

--001a114413c0d1c3bf0535651593
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>The attributes I provided are examples of attributes =
that might be provided in an access token; it was not meant to include all =
the attributes for this use case, because my intention was not to focus on =
conference server, but to discuss the idea in general.</div><div><br></div>=
<div><div>The following could be used with the conference factory to reques=
t a conference focus:</div><div>The INVITE sent to the conference factory c=
ould include a token that has a <b>conference-type</b> attribute which coul=
d take one of the following values: <b>audio, video, webconference, audio-w=
ebconference, or video-webconference</b>.</div><div>The conference factory =
will then issue a <b>conference uri</b> for the requested conference type, =
assuming the token is valid. If the <b>conference-type</b> attribute is mis=
sing, then the request would be rejected.</div></div><div><br></div><div>Is=
 that what you are looking for?</div><div><br></div><div>Regards,</div><div=
>=C2=A0Rifaat</div><div><br></div><div><br></div><div><br></div><div><br></=
div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Wed,=
 Jun 15, 2016 at 5:06 PM, Peterson, Jon <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:jon.peterson@neustar.biz" target=3D"_blank">jon.peterson@neustar.biz</=
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 style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif"><span class=3D"">
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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">
<span><br>
<br>
</span>
<div>
<div>In that case, if you could furnish a use case with attributes that wou=
ld be carried in a SIP iNVITE that are actually salient to the authorizatio=
n of that particular SIP request, we&#39;d be getting a lot closer to commo=
n ground.</div>
<div><br>
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>An obvious example would be an INVITE with a token sent to a conferenc=
e factory to authorize the creation of a focus.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span><div>We&#39;re still talking past each other. My point above was tha=
t the two attributes you specified - constraints on conference size and med=
ia types - have nothing to do with the authorization of the INVITE carrying=
 those attributes. Like, unless the conference
 size permitted is 0, then the INVITE that creates an ad hoc conference and=
 connects you to its bridge is not going to be rejected by the focus on the=
 basis of that conference size attribute. If the maximum conference size we=
re 0, the AuthZ service just wouldn&#39;t
 attach a token to an INVITE as it would never be allowed to start a confer=
ence. Same goes for media types.</div>
<div><br>
</div>
<div>Instead, what your two attributes constrain is what happens after the =
conference is created: when ten users try to dial in to a conference with a=
 permitted size of nine, say. That tenth INVITE would get rejected - not th=
e conference-forming INVITE that
 you propose would carry the token. That&#39;s why what you&#39;re proposin=
g is overloading the INVITE to tunnel conference policy constraints to the =
focus. I mean, maybe that&#39;s a problem space worth investigating, though=
 I suspect there are good reasons why RFC4353
 specified that conference policy is not manipulated with SIP - mostly beca=
use the semantics of manipulating conference policy probably don&#39;t map =
well onto dialog-forming SIP requests. But I think that conference policy p=
rovisioning problem space would be pretty
 far from where we started in this discussion.</div>
<div><br>
</div>
<div>Let me reiterate what I wrote in my previous mail, which I quote here =
at the top: &quot;furnish a use case with attributes that would be carried =
in a SIP iNVITE that are actually salient to the authorization of that part=
icular SIP request.&quot; Again: &quot;that particular
 SIP request.&quot; Not later transactions in some broader context in which=
 the original user agent probably isn&#39;t even an endpoint.</div>
<div><br>
</div>
<div>The Authorization header in SIP, for example, is actually used by the =
UAS to make a decision about whether or not to authorize the request in whi=
ch the header appears.=C2=A0 The token that you propose a SIP INVITE would =
carry in this case is not used by the
 relying party to make a decision about whether or not that INVITE is autho=
rized. That&#39;s why I believe this conference policy issue is a different=
 problem space entirely from figuring out how to implement attribute-based =
authorization for SIP requests.</div><span class=3D"">
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<div>Regards,</div>
<div>=C2=A0Rifaat</div>
<div><br>
</div>
</div>
</div>
</div>
</blockquote>
</span>
</span></div>

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

--001a114413c0d1c3bf0535651593--


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"Drage, Keith (Nokia - GB)" <keith.drage@nokia.com> writes:
> A reminder that RFC 3325 is an informational document, so I am not
> sure whether a standards track document can update an informational
> document. Anyone able to clarify?

I searched, but I couldn't find much guidance on what update documents
are or are not allowed to do in the BCP directory.

Dale


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Subject: [sipcore] Are 'alert' URNs being used?
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Does anyone know of implementations of the 'alert' URNs in Alert-Info
(RFC 7462) beyond the use of urn:alert:service:call-waiting by 3GPP?

Dale


From nobody Fri Jun 17 05:57:08 2016
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From: "Olle E. Johansson" <oej@edvina.net>
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Subject: [sipcore] SIPS must die - or at least be able to live without
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Friends,

I=E2=80=99ve tried to summarise my thoughts and the feedback in a =
presentation format.
=
http://www.slideshare.net/oej/sips-must-die-die-die-about-tls-usage-in-the=
-sip-protocol

Feedback is welcome and I=E2=80=99ll try to keep it up-to-date.

At this point, it=E2=80=99s not very well organised, as I have some more =
digging to do in regards
of SIP client certs and some other examples of TLS usage in related =
solutions like
MSRP, SIMPLE etc

It seems we are here:

- We have no agreement to remove SIPS: as a uri scheme, but agree that =
we need
  to be able to use TLS without it

- There are some thoughts on how to do this without outbound =
(half-outbound)

- We have not enough ideas on how to replace the SIPS requirement that =
all
   hops to the UA requiers TLS

- We have no suggestion on how to verify this

I return to S/MIME - what happened to that?

With the new thoughts about certificate handling used in WebRTC for DTLS =
and in Email for S/MIME using Dane
there may be some ideas to look at.

Personally I think removing SIPS: from the SIP set of standards would be =
a good thing. :-)

/O=


From nobody Fri Jun 17 06:15:57 2016
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From: "Olle E. Johansson" <oej@edvina.net>
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Date: Fri, 17 Jun 2016 15:15:50 +0200
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Subject: [sipcore] Half outbound
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http://www.slideshare.net/oej/sip-half-outbound-random-notes

Trying to summarize - am I right?

- No one seems to oppose some work in this area
- No one seems to say =E2=80=9CYay=E2=80=9D for Outbound and stop doing =
alternatives

As this is not an update to the core protocols, it seems like any draft =
that
can come out of this discussion needs to go to dispatch and follow
any route that group suggests.

Anyone that wants to join me in writing a first draft to start the =
process?

/O=


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From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Date: Fri, 17 Jun 2016 09:18:14 -0400
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Subject: Re: [sipcore] SIPS must die - or at least be able to live without
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--001a1142d222fda7aa0535792fed
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On Fri, Jun 17, 2016 at 8:57 AM, Olle E. Johansson <oej@edvina.net> wrote:

> Friends,
>
> I=E2=80=99ve tried to summarise my thoughts and the feedback in a present=
ation
> format.
>
> http://www.slideshare.net/oej/sips-must-die-die-die-about-tls-usage-in-th=
e-sip-protocol
>
> Feedback is welcome and I=E2=80=99ll try to keep it up-to-date.
>
> At this point, it=E2=80=99s not very well organised, as I have some more =
digging
> to do in regards
> of SIP client certs and some other examples of TLS usage in related
> solutions like
> MSRP, SIMPLE etc
>
> It seems we are here:
>
> - We have no agreement to remove SIPS: as a uri scheme, but agree that we
> need
>   to be able to use TLS without it
>

RFC5630 covers the use of SIP scheme over TLS already:
https://tools.ietf.org/html/rfc5630#section-3.1.3



>
> - There are some thoughts on how to do this without outbound
> (half-outbound)
>
> - We have not enough ideas on how to replace the SIPS requirement that al=
l
>    hops to the UA requiers TLS
>

Why would you do that?

Regards,
 Rifaat



> - We have no suggestion on how to verify this
>
> I return to S/MIME - what happened to that?
>
> With the new thoughts about certificate handling used in WebRTC for DTLS
> and in Email for S/MIME using Dane
> there may be some ideas to look at.
>
> Personally I think removing SIPS: from the SIP set of standards would be =
a
> good thing. :-)
>
> /O
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

--001a1142d222fda7aa0535792fed
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Fri, Jun 17, 2016 at 8:57 AM, Olle E. Johansson <span dir=3D"ltr">&l=
t;<a href=3D"mailto:oej@edvina.net" target=3D"_blank">oej@edvina.net</a>&gt=
;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0p=
x 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-color=
:rgb(204,204,204);padding-left:1ex">Friends,<br>
<br>
I=E2=80=99ve tried to summarise my thoughts and the feedback in a presentat=
ion format.<br>
<a href=3D"http://www.slideshare.net/oej/sips-must-die-die-die-about-tls-us=
age-in-the-sip-protocol" rel=3D"noreferrer" target=3D"_blank">http://www.sl=
ideshare.net/oej/sips-must-die-die-die-about-tls-usage-in-the-sip-protocol<=
/a><br>
<br>
Feedback is welcome and I=E2=80=99ll try to keep it up-to-date.<br>
<br>
At this point, it=E2=80=99s not very well organised, as I have some more di=
gging to do in regards<br>
of SIP client certs and some other examples of TLS usage in related solutio=
ns like<br>
MSRP, SIMPLE etc<br>
<br>
It seems we are here:<br>
<br>
- We have no agreement to remove SIPS: as a uri scheme, but agree that we n=
eed<br>
=C2=A0 to be able to use TLS without it<br></blockquote><div><br></div><div=
>RFC5630 covers the use of SIP scheme over TLS already:</div><div><a href=
=3D"https://tools.ietf.org/html/rfc5630#section-3.1.3">https://tools.ietf.o=
rg/html/rfc5630#section-3.1.3</a><br></div><div><br></div><div>=C2=A0</div>=
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);p=
adding-left:1ex">
<br>
- There are some thoughts on how to do this without outbound (half-outbound=
)<br>
<br>
- We have not enough ideas on how to replace the SIPS requirement that all<=
br>
=C2=A0 =C2=A0hops to the UA requiers TLS<br></blockquote><div><br></div><di=
v>Why would you do that?</div><div><br></div><div>Regards,</div><div>=C2=A0=
Rifaat</div><div>=C2=A0</div><div><br></div><blockquote class=3D"gmail_quot=
e" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-styl=
e:solid;border-left-color:rgb(204,204,204);padding-left:1ex">
<br>
- We have no suggestion on how to verify this<br>
<br>
I return to S/MIME - what happened to that?<br>
<br>
With the new thoughts about certificate handling used in WebRTC for DTLS an=
d in Email for S/MIME using Dane<br>
there may be some ideas to look at.<br>
<br>
Personally I think removing SIPS: from the SIP set of standards would be a =
good thing. :-)<br>
<br>
/O<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" rel=3D"noreferrer=
" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sipcore</a><br>
</blockquote></div><br></div></div>

--001a1142d222fda7aa0535792fed--


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To: "Olle E. Johansson" <oej@edvina.net>, SIPCORE <sipcore@ietf.org>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/sipcore/hQU33oEAiOZsWHv7-aKEvUqFnRE>
Subject: Re: [sipcore] Half outbound
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From: "Olle E. Johansson" <oej@edvina.net>
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To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
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Cc: SIPCORE <sipcore@ietf.org>, Olle E Johansson <oej@edvina.net>
Subject: Re: [sipcore] SIPS must die - or at least be able to live without
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> On 17 Jun 2016, at 15:18, Rifaat Shekh-Yusef <rifaat.ietf@gmail.com> =
wrote:
>=20
>=20
>=20
> On Fri, Jun 17, 2016 at 8:57 AM, Olle E. Johansson <oej@edvina.net =
<mailto:oej@edvina.net>> wrote:
> Friends,
>=20
> I=E2=80=99ve tried to summarise my thoughts and the feedback in a =
presentation format.
> =
http://www.slideshare.net/oej/sips-must-die-die-die-about-tls-usage-in-the=
-sip-protocol =
<http://www.slideshare.net/oej/sips-must-die-die-die-about-tls-usage-in-th=
e-sip-protocol>
>=20
> Feedback is welcome and I=E2=80=99ll try to keep it up-to-date.
>=20
> At this point, it=E2=80=99s not very well organised, as I have some =
more digging to do in regards
> of SIP client certs and some other examples of TLS usage in related =
solutions like
> MSRP, SIMPLE etc
>=20
> It seems we are here:
>=20
> - We have no agreement to remove SIPS: as a uri scheme, but agree that =
we need
>   to be able to use TLS without it
>=20
> RFC5630 covers the use of SIP scheme over TLS already:
> https://tools.ietf.org/html/rfc5630#section-3.1.3 =
<https://tools.ietf.org/html/rfc5630#section-3.1.3>
>=20
And leave some gaps that I documented in my presentation=E2=80=A6 As I =
say on one slide
this RFC points to RFC 5626 but no one seems to want to implement it so =
it=E2=80=99s not
the solution. Check the presentation!

> =20
>=20
> - There are some thoughts on how to do this without outbound =
(half-outbound)
>=20
> - We have not enough ideas on how to replace the SIPS requirement that =
all
>    hops to the UA requiers TLS
>=20
> Why would you do that?
Well, some people on this list pointed that out as a feature they wanted
when I said that I just wanted to remove it so I=E2=80=99ll let them =
respond.

/O
>=20
> Regards,
>  Rifaat
> =20
>=20
>=20
> - We have no suggestion on how to verify this
>=20
> I return to S/MIME - what happened to that?
>=20
> With the new thoughts about certificate handling used in WebRTC for =
DTLS and in Email for S/MIME using Dane
> there may be some ideas to look at.
>=20
> Personally I think removing SIPS: from the SIP set of standards would =
be a good thing. :-)
>=20
> /O
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org <mailto:sipcore@ietf.org>
> https://www.ietf.org/mailman/listinfo/sipcore =
<https://www.ietf.org/mailman/listinfo/sipcore>
>=20


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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; -webkit-line-break: after-white-space;" =
class=3D""><br class=3D""><div><blockquote type=3D"cite" class=3D""><div =
class=3D"">On 17 Jun 2016, at 15:18, Rifaat Shekh-Yusef &lt;<a =
href=3D"mailto:rifaat.ietf@gmail.com" =
class=3D"">rifaat.ietf@gmail.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><div dir=3D"ltr" =
class=3D""><br class=3D""><div class=3D"gmail_extra"><br class=3D""><div =
class=3D"gmail_quote">On Fri, Jun 17, 2016 at 8:57 AM, Olle E. Johansson =
<span dir=3D"ltr" class=3D"">&lt;<a href=3D"mailto:oej@edvina.net" =
target=3D"_blank" class=3D"">oej@edvina.net</a>&gt;</span> wrote:<br =
class=3D""><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px =
0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb(=
204,204,204);padding-left:1ex">Friends,<br class=3D"">
<br class=3D"">
I=E2=80=99ve tried to summarise my thoughts and the feedback in a =
presentation format.<br class=3D"">
<a =
href=3D"http://www.slideshare.net/oej/sips-must-die-die-die-about-tls-usag=
e-in-the-sip-protocol" rel=3D"noreferrer" target=3D"_blank" =
class=3D"">http://www.slideshare.net/oej/sips-must-die-die-die-about-tls-u=
sage-in-the-sip-protocol</a><br class=3D"">
<br class=3D"">
Feedback is welcome and I=E2=80=99ll try to keep it up-to-date.<br =
class=3D"">
<br class=3D"">
At this point, it=E2=80=99s not very well organised, as I have some more =
digging to do in regards<br class=3D"">
of SIP client certs and some other examples of TLS usage in related =
solutions like<br class=3D"">
MSRP, SIMPLE etc<br class=3D"">
<br class=3D"">
It seems we are here:<br class=3D"">
<br class=3D"">
- We have no agreement to remove SIPS: as a uri scheme, but agree that =
we need<br class=3D"">
&nbsp; to be able to use TLS without it<br class=3D""></blockquote><div =
class=3D""><br class=3D""></div><div class=3D"">RFC5630 covers the use =
of SIP scheme over TLS already:</div><div class=3D""><a =
href=3D"https://tools.ietf.org/html/rfc5630#section-3.1.3" =
class=3D"">https://tools.ietf.org/html/rfc5630#section-3.1.3</a><br =
class=3D""></div><div class=3D""><br =
class=3D""></div></div></div></div></div></blockquote>And leave some =
gaps that I documented in my presentation=E2=80=A6 As I say on one =
slide</div><div>this RFC points to RFC 5626 but no one seems to want to =
implement it so it=E2=80=99s not</div><div>the solution. Check the =
presentation!</div><div><br class=3D""><blockquote type=3D"cite" =
class=3D""><div class=3D""><div dir=3D"ltr" class=3D""><div =
class=3D"gmail_extra"><div class=3D"gmail_quote"><div =
class=3D"">&nbsp;</div><blockquote class=3D"gmail_quote" =
style=3D"margin:0px 0px 0px =
0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb(=
204,204,204);padding-left:1ex">
<br class=3D"">
- There are some thoughts on how to do this without outbound =
(half-outbound)<br class=3D"">
<br class=3D"">
- We have not enough ideas on how to replace the SIPS requirement that =
all<br class=3D"">
&nbsp; &nbsp;hops to the UA requiers TLS<br class=3D""></blockquote><div =
class=3D""><br class=3D""></div><div class=3D"">Why would you do =
that?</div></div></div></div></div></blockquote>Well, some people on =
this list pointed that out as a feature they wanted</div><div>when I =
said that I just wanted to remove it so I=E2=80=99ll let them =
respond.</div><div><br class=3D""></div><div>/O<br class=3D""><blockquote =
type=3D"cite" class=3D""><div class=3D""><div dir=3D"ltr" class=3D""><div =
class=3D"gmail_extra"><div class=3D"gmail_quote"><div class=3D""><br =
class=3D""></div><div class=3D"">Regards,</div><div =
class=3D"">&nbsp;Rifaat</div><div class=3D"">&nbsp;</div><div =
class=3D""><br class=3D""></div><blockquote class=3D"gmail_quote" =
style=3D"margin:0px 0px 0px =
0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb(=
204,204,204);padding-left:1ex">
<br class=3D"">
- We have no suggestion on how to verify this<br class=3D"">
<br class=3D"">
I return to S/MIME - what happened to that?<br class=3D"">
<br class=3D"">
With the new thoughts about certificate handling used in WebRTC for DTLS =
and in Email for S/MIME using Dane<br class=3D"">
there may be some ideas to look at.<br class=3D"">
<br class=3D"">
Personally I think removing SIPS: from the SIP set of standards would be =
a good thing. :-)<br class=3D"">
<br class=3D"">
/O<br class=3D"">
_______________________________________________<br class=3D"">
sipcore mailing list<br class=3D"">
<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank" =
class=3D"">sipcore@ietf.org</a><br class=3D"">
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" =
rel=3D"noreferrer" target=3D"_blank" =
class=3D"">https://www.ietf.org/mailman/listinfo/sipcore</a><br =
class=3D"">
</blockquote></div><br class=3D""></div></div>
</div></blockquote></div><br class=3D""></body></html>=

--Apple-Mail=_9EB03A36-8757-4C0C-B663-8F7C58D965F2--


From nobody Fri Jun 17 06:22:39 2016
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To: "Neelakantan, Neel" <nneelakantan@sonusnet.com>
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Subject: Re: [sipcore] Half outbound
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> On 17 Jun 2016, at 15:18, Neelakantan, Neel =
<nneelakantan@sonusnet.com> wrote:
>=20
> Olle,
>=20
> I am willing to help and contribute in this.  I know we discussed this =
in SIP Forums when the outbound came up.
Great! Welcome to the team :-)

I=E2=80=99ll start with writing an outline and committing to github and =
we=E2=80=99ll can take it from there.

Cheers,
/O
>=20
> Thanks,
> Neel.
>=20
>> -----Original Message-----
>> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Olle E.
>> Johansson
>> Sent: Friday, June 17, 2016 8:16 AM
>> To: SIPCORE <sipcore@ietf.org>
>> Cc: Olle E Johansson <oej@edvina.net>
>> Subject: [sipcore] Half outbound
>>=20
>> http://www.slideshare.net/oej/sip-half-outbound-random-notes
>>=20
>> Trying to summarize - am I right?
>>=20
>> - No one seems to oppose some work in this area
>> - No one seems to say =E2=80=9CYay=E2=80=9D for Outbound and stop =
doing alternatives
>>=20
>> As this is not an update to the core protocols, it seems like any =
draft that can
>> come out of this discussion needs to go to dispatch and follow any =
route that
>> group suggests.
>>=20
>> Anyone that wants to join me in writing a first draft to start the =
process?
>>=20
>> /O
>> _______________________________________________
>> 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 Fri Jun 17 07:48:41 2016
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To: sipcore@ietf.org
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] Half outbound
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On 6/17/16 9:15 AM, Olle E. Johansson wrote:
> http://www.slideshare.net/oej/sip-half-outbound-random-notes
>
> Trying to summarize - am I right?
>
> - No one seems to oppose some work in this area
> - No one seems to say â€œYayâ€ for Outbound and stop doing alternatives
>
> As this is not an update to the core protocols, it seems like any draft that
> can come out of this discussion needs to go to dispatch and follow
> any route that group suggests.
>
> Anyone that wants to join me in writing a first draft to start the process?

I thought we had concluded that half-outbound isn't needed, because the 
existing outbound may be used with a single connection/flow if properly 
configured.

If people still don't want to do it, then maybe what we need is research 
into why.

	Thanks,
	Paul


From nobody Fri Jun 17 07:52:28 2016
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> On 17 Jun 2016, at 16:48, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>=20
> On 6/17/16 9:15 AM, Olle E. Johansson wrote:
>> http://www.slideshare.net/oej/sip-half-outbound-random-notes
>>=20
>> Trying to summarize - am I right?
>>=20
>> - No one seems to oppose some work in this area
>> - No one seems to say =E2=80=9CYay=E2=80=9D for Outbound and stop =
doing alternatives
>>=20
>> As this is not an update to the core protocols, it seems like any =
draft that
>> can come out of this discussion needs to go to dispatch and follow
>> any route that group suggests.
>>=20
>> Anyone that wants to join me in writing a first draft to start the =
process?
>=20
> I thought we had concluded that half-outbound isn't needed, because =
the existing outbound may be used with a single connection/flow if =
properly configured.
>=20
> If people still don't want to do it, then maybe what we need is =
research into why.

The SIPconnect team more or less refused to add outbound regardless on
how many times I pointed at it as the only solution. I hope they can =
answer!

So let=E2=80=99s take another round :-)


/O=


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Subject: Re: [sipcore] SIPS must die - or at least be able to live without
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On 6/17/16 8:57 AM, Olle E. Johansson wrote:
> Friends,
>
> Iâ€™ve tried to summarise my thoughts and the feedback in a presentation format.
> http://www.slideshare.net/oej/sips-must-die-die-die-about-tls-usage-in-the-sip-protocol
>
> Feedback is welcome and Iâ€™ll try to keep it up-to-date.
>
> At this point, itâ€™s not very well organised, as I have some more digging to do in regards
> of SIP client certs and some other examples of TLS usage in related solutions like
> MSRP, SIMPLE etc
>
> It seems we are here:
>
> - We have no agreement to remove SIPS: as a uri scheme, but agree that we need
>   to be able to use TLS without it
>
> - There are some thoughts on how to do this without outbound (half-outbound)
>
> - We have not enough ideas on how to replace the SIPS requirement that all
>    hops to the UA requiers TLS
>
> - We have no suggestion on how to verify this
>
> I return to S/MIME - what happened to that?

I remember back when I was first learning about SIP (with RFC2543) I 
spent some effort trying to understand the S/MIME stuff. I was never 
able to figure out how it could possibly work. After asking a few 
questions I finally stopped thinking about it.

> With the new thoughts about certificate handling used in WebRTC for DTLS and in Email for S/MIME using Dane
> there may be some ideas to look at.
>
> Personally I think removing SIPS: from the SIP set of standards would be a good thing. :-)

The one thing SIPS offers is a way to indicate a *requirement* by the 
UAC for security on every hop. Unfortunately the realization of that 
hasn't worked out.

I suppose the same thing could be achieved using Require and 
Proxy-Require. But still there is no obvious way to verify it has been 
honored.

I think I recall at one time there was a proposal for how to establish 
an e2e TLS association. But nothing came of it. Maybe something could be 
worked out that way. For instance, if webrtc can succeed in establishing 
an e2d DTLS connection for media, why can't the same be done for sip? 
And if it can be done for media, why couldn't it also be done for 
signaling?

For instance, use SIP to establish a session, with ICE used to establish 
an RTCWEB data channel. Then do further signaling over the data channel. 
Use that to establish other media streams.

	Thanks,
	Paul


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Subject: Re: [sipcore] SIPS must die - or at least be able to live without
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> On 17 Jun 2016, at 17:13, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>=20
> On 6/17/16 8:57 AM, Olle E. Johansson wrote:
>> Friends,
>>=20
>> I=E2=80=99ve tried to summarise my thoughts and the feedback in a =
presentation format.
>> =
http://www.slideshare.net/oej/sips-must-die-die-die-about-tls-usage-in-the=
-sip-protocol
>>=20
>> Feedback is welcome and I=E2=80=99ll try to keep it up-to-date.
>>=20
>> At this point, it=E2=80=99s not very well organised, as I have some =
more digging to do in regards
>> of SIP client certs and some other examples of TLS usage in related =
solutions like
>> MSRP, SIMPLE etc
>>=20
>> It seems we are here:
>>=20
>> - We have no agreement to remove SIPS: as a uri scheme, but agree =
that we need
>>  to be able to use TLS without it
>>=20
>> - There are some thoughts on how to do this without outbound =
(half-outbound)
>>=20
>> - We have not enough ideas on how to replace the SIPS requirement =
that all
>>   hops to the UA requiers TLS
>>=20
>> - We have no suggestion on how to verify this
>>=20
>> I return to S/MIME - what happened to that?
>=20
> I remember back when I was first learning about SIP (with RFC2543) I =
spent some effort trying to understand the S/MIME stuff. I was never =
able to figure out how it could possibly work. After asking a few =
questions I finally stopped thinking about it.
Oh yeah, I=E2=80=99ve been struggling with RSA lawyers and understanding =
it a long time ago. The difference is that there=E2=80=99s
a bit more code out there now and a little - although not much - more =
experience in usage of it e-mail.

>=20
>> With the new thoughts about certificate handling used in WebRTC for =
DTLS and in Email for S/MIME using Dane
>> there may be some ideas to look at.
>>=20
>> Personally I think removing SIPS: from the SIP set of standards would =
be a good thing. :-)
>=20
> The one thing SIPS offers is a way to indicate a *requirement* by the =
UAC for security on every hop. Unfortunately the realization of that =
hasn't worked out.
>=20
> I suppose the same thing could be achieved using Require and =
Proxy-Require. But still there is no obvious way to verify it has been =
honored.
>=20
> I think I recall at one time there was a proposal for how to establish =
an e2e TLS association. But nothing came of it.
Yep, some sort of websocket-like upgrade - I do remember there was such =
a draft too.

> Maybe something could be worked out that way. For instance, if webrtc =
can succeed in establishing an e2d DTLS connection for media, why can't =
the same be done for sip? And if it can be done for media, why couldn't =
it also be done for signaling?
>=20
> For instance, use SIP to establish a session, with ICE used to =
establish an RTCWEB data channel. Then do further signaling over the =
data channel. Use that to establish other media streams.
That=E2=80=99s one idea :-)

/O=


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To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: [sipcore] SIPS must die - or at least be able to live without
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Subject: Re: [sipcore] SIPS must die - or at least be able to live without
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Subject: Re: [sipcore] SIPS must die - or at least be able to live without
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On 6/17/16 4:03 PM, Christer Holmberg wrote:
> Hi,
>
> ...
>
>> I think I recall at one time there was a proposal for how to establish an e2e TLS association. But nothing came of it. Maybe something could be worked out that way. For instance, if webrtc can succeed in >establishing an e2d DTLS connection for media, why can't the same be done for sip?
>> And if it can be done for media, why couldn't it also be done for signaling?
>
> Because SIP is not an e2e protocol: intermediaries need to have access (read: decrypt) signalling information.
>
>> For instance, use SIP to establish a session, with ICE used to establish an RTCWEB data channel. Then do further signaling over the data channel.
>> Use that to establish other media streams.
>
> ...or, as it it's also known as: H.245 :)
>
> But, the question is WHAT information you want to protect.

Yes, that is important to decide.

Normal users most likely want media protected, at least.

Next, I suppose many people would like "call metadata" protected. Almost 
nothing we will do will protect that. I guess SIP over TOR perhaps. 
(Though it seems TOR isn't safe either.)

	Thanks,
	Paul


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From: Roman Shpount <roman@telurix.com>
Date: Fri, 17 Jun 2016 18:08:19 -0400
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To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] SIPS must die - or at least be able to live without
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--94eb2c114246aaaf9e0535809788
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On Fri, Jun 17, 2016 at 5:39 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> On 6/17/16 4:03 PM, Christer Holmberg wrote:
>
>> But, the question is WHAT information you want to protect.
>>
>
> Yes, that is important to decide.
>
> Normal users most likely want media protected, at least.
>

The best way to protect media is DTLS-SRTP. The media will be protected
even if signaling is compromised (as long as it is correctly authenticated).


> Next, I suppose many people would like "call metadata" protected. Almost
> nothing we will do will protect that. I guess SIP over TOR perhaps. (Though
> it seems TOR isn't safe either.)
>

There is also the issue of end-to-end authentication of signaling. It is
not only important to protect the information but it is also important to
make sure that session description and the user identity was not tempered
with. In some sense this problem is exactly the same as identity issues
which are part of WebRTC.

I guess the most important issue I see with this discussion is definition
of scope. SIP can be the end point communication protocol or federation
protocol between service providers. SIP over TLS with certificate based
authentication is perfect for communicating between different services
located on public IP (federation). It will work quite well if you need to
terminate a SIP call to a PSTN provider or accept a PSTN call to a VoIP
service.

SIP is a horrible protocol for end points with or without TLS. It tries to
enable the operations of 12-button VoIP phones, but manages to be broken
for almost any feature. Security, NAT traversal, presence, distinctive
ringing, MWI, etc all seems to be broken in almost all practical
implementations. I hope these ancient phones will die together with SIP on
the end points and get replaced by WebRTC based implementations. I just do
not see why would anybody at this point build a new SIP end point device
instead of making something browser based. At this point I would use a push
service, such as FireBase Messaging (
https://firebase.google.com/docs/cloud-messaging/) instead of registrations
to notify about incoming calls or presence related updates, WebRTC for real
time media and dynamic HTML for UI. This will be easier to build, will
provide better feature set, scale better, be a lot more secure and use less
power on mobile.

Regards,
_____________
Roman Shpount

--94eb2c114246aaaf9e0535809788
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure" data-smartmail=3D"gmail_signature">On Fri, Jun 17, 2016 at 5:39 PM, Pa=
ul Kyzivat <span dir=3D"ltr">&lt;<a href=3D"mailto:pkyzivat@alum.mit.edu" t=
arget=3D"_blank">pkyzivat@alum.mit.edu</a>&gt;</span> wrote:<br></div></div=
><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"marg=
in:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1e=
x">On 6/17/16 4:03 PM, Christer Holmberg wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left:1px solid rgb(204,204,204);padding-left:1ex">But, the question is WHAT=
 information you want to protect.<br>
</blockquote>
<br>
Yes, that is important to decide.<br>
<br>
Normal users most likely want media protected, at least.<br></blockquote><d=
iv><br></div><div>The best way to protect media is DTLS-SRTP. The media wil=
l be protected even if signaling is compromised (as long as it is correctly=
 authenticated).</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padd=
ing-left:1ex">Next, I suppose many people would like &quot;call metadata&qu=
ot; protected. Almost nothing we will do will protect that. I guess SIP ove=
r TOR perhaps. (Though it seems TOR isn&#39;t safe either.)<br></blockquote=
><div><br></div><div>There is also the issue of end-to-end authentication o=
f signaling. It is not only important to protect the information but it is =
also important to make sure that session description and the user identity =
was not tempered with. In some sense this problem is exactly the same as id=
entity issues which are part of WebRTC.</div><div><br></div><div>I guess th=
e most important issue I see with this discussion is definition of scope. S=
IP can be the end point communication protocol or federation protocol betwe=
en service providers. SIP over TLS with certificate based authentication is=
 perfect for communicating between different services located on public IP =
(federation). It will work quite well if you need to terminate a SIP call t=
o a PSTN provider or accept a PSTN call to a VoIP service.</div><div><br></=
div><div>SIP is a horrible protocol for end points with or without TLS. It =
tries to enable the operations of 12-button VoIP phones, but manages to be =
broken for almost any feature. Security, NAT traversal, presence, distincti=
ve ringing, MWI, etc all seems to be broken in almost all practical impleme=
ntations. I hope these ancient phones will die together with SIP on the end=
 points and get replaced by WebRTC based implementations. I just do not see=
 why would anybody at this point build a new SIP end point device instead o=
f making something browser based. At this point I would use a push service,=
 such as FireBase Messaging (<a href=3D"https://firebase.google.com/docs/cl=
oud-messaging/">https://firebase.google.com/docs/cloud-messaging/</a>) inst=
ead of registrations to notify about incoming calls or presence related upd=
ates, WebRTC for real time media and dynamic HTML for UI. This will be easi=
er to build, will provide better feature set, scale better, be a lot more s=
ecure and use less power on mobile.</div><div><br></div><div>Regards,</div>=
<div><div class=3D"gmail_signature" data-smartmail=3D"gmail_signature">____=
_________<br>Roman Shpount</div></div><div>=C2=A0</div></div></div></div>

--94eb2c114246aaaf9e0535809788--


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From: "Asveren, Tolga" <tasveren@sonusnet.com>
To: Roman Shpount <roman@telurix.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [sipcore] SIPS must die - or at least be able to live without
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Olle E. Johansson" <oej@edvina.net>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [sipcore] Half outbound
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Subject: Re: [sipcore] SIPS must die - or at least be able to live without
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On 6/17/16 11:31 PM, Asveren, Tolga wrote:
> â€œThere is also the issue of end-to-end authentication of signaling. It
> is not only important to protect the information but it is also
> important to make sure that session description and the user identity
> was not tempered with. In some sense this problem is exactly the same as
> identity issues which are part of WebRTC.â€
>
>
>
> You hit the nail on the head. The assumption of â€œsecure end-to-end
> signalingâ€ (use of fingerprint attribute)  is a prerequisite for
> DTLS-SRTP to be meaningful from security perspective.

For the common telephony use cases it is going to be tricky to figure 
out what we mean here. It only really works e2e if the calling and 
called endpoints possess unique credentials known to both. Otherwise, 
one way or another you are relying on transitive trust.

For instance, when I call the Acme Bank, I probably don't know exactly 
who I am calling or have a credential for them. Every worker at the bank 
could have the bank's credential that I know, but that is difficult to 
administer. Otherwise I'm relying on transitive trust between a server 
at the bank that has those credentials, and the worker's UA.

Note that Webrtc relies on transitive trust between the two ends to 
relay the fingerprints, via at least one server in the middle. And just 
like with SIP, there is no guarantee that signaling on each leg is 
protected.

	Thanks,
	Paul


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From: "Olle E. Johansson" <oej@edvina.net>
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Date: Sat, 18 Jun 2016 21:45:09 +0200
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [sipcore] Half outbound
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> On 18 Jun 2016, at 11:02, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>=20
>=20
> Hi,
>=20
> If you plan to bring this to DISPATCH, I suggest the following:
>=20
> 1. Don't call it "Half Outbound". It will people focus on the =
mechanism instead of what you actually want to achieve.
>=20
> 2. Write down a high-level use-case/call flow. Start by showing people =
what you want to do - not by showing what is wrong with existing =
mechanism (Outbound etc).
>=20
> 3. Based on the use-case/call-flow, write down a list of requirements. =
Again, leave the SIP protocol mechanism specifics out.
>=20
> 4. THEN you can start talking about protocol mechanisms. You can e.g. =
pick the parts of outbound that would fulfil the requirements. THAT you =
can call "half outbound".
>=20
> 5. If you are going to make statements like "people don't want to =
implement outbound", you really need some justification and background =
information. Otherwise people will just ask "how do you know people will =
implement this new mechanism?=E2=80=9D.

Great feedback, thank you!

/Olle

>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
> -----Original Message-----
> From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Olle E. =
Johansson
> Sent: 17 June 2016 17:52
> To: Paul Kyzivat <pkyzivat@alum.mit.edu>
> Cc: sipcore@ietf.org; Olle E Johansson <oej@edvina.net>
> Subject: Re: [sipcore] Half outbound
>=20
>=20
>> On 17 Jun 2016, at 16:48, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>>=20
>> On 6/17/16 9:15 AM, Olle E. Johansson wrote:
>>> http://www.slideshare.net/oej/sip-half-outbound-random-notes
>>>=20
>>> Trying to summarize - am I right?
>>>=20
>>> - No one seems to oppose some work in this area
>>> - No one seems to say =E2=80=9CYay=E2=80=9D for Outbound and stop =
doing alternatives
>>>=20
>>> As this is not an update to the core protocols, it seems like any=20
>>> draft that can come out of this discussion needs to go to dispatch=20=

>>> and follow any route that group suggests.
>>>=20
>>> Anyone that wants to join me in writing a first draft to start the =
process?
>>=20
>> I thought we had concluded that half-outbound isn't needed, because =
the existing outbound may be used with a single connection/flow if =
properly configured.
>>=20
>> If people still don't want to do it, then maybe what we need is =
research into why.
>=20
> The SIPconnect team more or less refused to add outbound regardless on =
how many times I pointed at it as the only solution. I hope they can =
answer!
>=20
> So let=E2=80=99s take another round :-)
>=20
>=20
> /O
> _______________________________________________
> 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: "Asveren, Tolga" <tasveren@sonusnet.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, Roman Shpount <roman@telurix.com>
Thread-Topic: [sipcore] SIPS must die - or at least be able to live without
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From: Ranjit Avasarala <ranjitkav0811@gmail.com>
Date: Sat, 18 Jun 2016 23:23:06 -0500
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To: "Olle E. Johansson" <oej@edvina.net>
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Subject: Re: [sipcore] Half outbound
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--001a1141efced3d1d1053599f1a6
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yes I agree - half outbound as a name does not sound appropriate - a more
meaningful name is better

Regards
Ranjit

On Sat, Jun 18, 2016 at 2:45 PM, Olle E. Johansson <oej@edvina.net> wrote:

>
> > On 18 Jun 2016, at 11:02, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
> >
> >
> > Hi,
> >
> > If you plan to bring this to DISPATCH, I suggest the following:
> >
> > 1. Don't call it "Half Outbound". It will people focus on the mechanism
> instead of what you actually want to achieve.
> >
> > 2. Write down a high-level use-case/call flow. Start by showing people
> what you want to do - not by showing what is wrong with existing mechanis=
m
> (Outbound etc).
> >
> > 3. Based on the use-case/call-flow, write down a list of requirements.
> Again, leave the SIP protocol mechanism specifics out.
> >
> > 4. THEN you can start talking about protocol mechanisms. You can e.g.
> pick the parts of outbound that would fulfil the requirements. THAT you c=
an
> call "half outbound".
> >
> > 5. If you are going to make statements like "people don't want to
> implement outbound", you really need some justification and background
> information. Otherwise people will just ask "how do you know people will
> implement this new mechanism?=E2=80=9D.
>
> Great feedback, thank you!
>
> /Olle
>
> >
> > Regards,
> >
> > Christer
> >
> >
> >
> > -----Original Message-----
> > From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Olle E.
> Johansson
> > Sent: 17 June 2016 17:52
> > To: Paul Kyzivat <pkyzivat@alum.mit.edu>
> > Cc: sipcore@ietf.org; Olle E Johansson <oej@edvina.net>
> > Subject: Re: [sipcore] Half outbound
> >
> >
> >> On 17 Jun 2016, at 16:48, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
> >>
> >> On 6/17/16 9:15 AM, Olle E. Johansson wrote:
> >>> http://www.slideshare.net/oej/sip-half-outbound-random-notes
> >>>
> >>> Trying to summarize - am I right?
> >>>
> >>> - No one seems to oppose some work in this area
> >>> - No one seems to say =E2=80=9CYay=E2=80=9D for Outbound and stop doi=
ng alternatives
> >>>
> >>> As this is not an update to the core protocols, it seems like any
> >>> draft that can come out of this discussion needs to go to dispatch
> >>> and follow any route that group suggests.
> >>>
> >>> Anyone that wants to join me in writing a first draft to start the
> process?
> >>
> >> I thought we had concluded that half-outbound isn't needed, because th=
e
> existing outbound may be used with a single connection/flow if properly
> configured.
> >>
> >> If people still don't want to do it, then maybe what we need is
> research into why.
> >
> > The SIPconnect team more or less refused to add outbound regardless on
> how many times I pointed at it as the only solution. I hope they can answ=
er!
> >
> > So let=E2=80=99s take another round :-)
> >
> >
> > /O
> > _______________________________________________
> > 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
>

--001a1141efced3d1d1053599f1a6
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<div dir=3D"ltr">yes I agree - half outbound as a name does not sound appro=
priate - a more meaningful name is better<div><br></div><div>Regards</div><=
div>Ranjit</div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_qu=
ote">On Sat, Jun 18, 2016 at 2:45 PM, Olle E. Johansson <span dir=3D"ltr">&=
lt;<a href=3D"mailto:oej@edvina.net" target=3D"_blank">oej@edvina.net</a>&g=
t;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0=
 .8ex;border-left:1px #ccc solid;padding-left:1ex"><br>
&gt; On 18 Jun 2016, at 11:02, Christer Holmberg &lt;<a href=3D"mailto:chri=
ster.holmberg@ericsson.com">christer.holmberg@ericsson.com</a>&gt; wrote:<b=
r>
&gt;<br>
&gt;<br>
&gt; Hi,<br>
&gt;<br>
&gt; If you plan to bring this to DISPATCH, I suggest the following:<br>
&gt;<br>
&gt; 1. Don&#39;t call it &quot;Half Outbound&quot;. It will people focus o=
n the mechanism instead of what you actually want to achieve.<br>
&gt;<br>
&gt; 2. Write down a high-level use-case/call flow. Start by showing people=
 what you want to do - not by showing what is wrong with existing mechanism=
 (Outbound etc).<br>
&gt;<br>
&gt; 3. Based on the use-case/call-flow, write down a list of requirements.=
 Again, leave the SIP protocol mechanism specifics out.<br>
&gt;<br>
&gt; 4. THEN you can start talking about protocol mechanisms. You can e.g. =
pick the parts of outbound that would fulfil the requirements. THAT you can=
 call &quot;half outbound&quot;.<br>
&gt;<br>
&gt; 5. If you are going to make statements like &quot;people don&#39;t wan=
t to implement outbound&quot;, you really need some justification and backg=
round information. Otherwise people will just ask &quot;how do you know peo=
ple will implement this new mechanism?=E2=80=9D.<br>
<br>
Great feedback, thank you!<br>
<br>
/Olle<br>
<br>
&gt;<br>
&gt; Regards,<br>
&gt;<br>
&gt; Christer<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; From: sipcore [mailto:<a href=3D"mailto:sipcore-bounces@ietf.org">sipc=
ore-bounces@ietf.org</a>] On Behalf Of Olle E. Johansson<br>
&gt; Sent: 17 June 2016 17:52<br>
&gt; To: Paul Kyzivat &lt;<a href=3D"mailto:pkyzivat@alum.mit.edu">pkyzivat=
@alum.mit.edu</a>&gt;<br>
&gt; Cc: <a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a>; Olle E J=
ohansson &lt;<a href=3D"mailto:oej@edvina.net">oej@edvina.net</a>&gt;<br>
&gt; Subject: Re: [sipcore] Half outbound<br>
&gt;<br>
&gt;<br>
&gt;&gt; On 17 Jun 2016, at 16:48, Paul Kyzivat &lt;<a href=3D"mailto:pkyzi=
vat@alum.mit.edu">pkyzivat@alum.mit.edu</a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt; On 6/17/16 9:15 AM, Olle E. Johansson wrote:<br>
&gt;&gt;&gt; <a href=3D"http://www.slideshare.net/oej/sip-half-outbound-ran=
dom-notes" rel=3D"noreferrer" target=3D"_blank">http://www.slideshare.net/o=
ej/sip-half-outbound-random-notes</a><br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Trying to summarize - am I right?<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; - No one seems to oppose some work in this area<br>
&gt;&gt;&gt; - No one seems to say =E2=80=9CYay=E2=80=9D for Outbound and s=
top doing alternatives<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; As this is not an update to the core protocols, it seems like =
any<br>
&gt;&gt;&gt; draft that can come out of this discussion needs to go to disp=
atch<br>
&gt;&gt;&gt; and follow any route that group suggests.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Anyone that wants to join me in writing a first draft to start=
 the process?<br>
&gt;&gt;<br>
&gt;&gt; I thought we had concluded that half-outbound isn&#39;t needed, be=
cause the existing outbound may be used with a single connection/flow if pr=
operly configured.<br>
&gt;&gt;<br>
&gt;&gt; If people still don&#39;t want to do it, then maybe what we need i=
s research into why.<br>
&gt;<br>
&gt; The SIPconnect team more or less refused to add outbound regardless on=
 how many times I pointed at it as the only solution. I hope they can answe=
r!<br>
&gt;<br>
&gt; So let=E2=80=99s take another round :-)<br>
&gt;<br>
&gt;<br>
&gt; /O<br>
&gt; _______________________________________________<br>
&gt; sipcore mailing list<br>
&gt; <a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" rel=3D"noref=
errer" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sipcore</a><=
br>
&gt; _______________________________________________<br>
&gt; sipcore mailing list<br>
&gt; <a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" rel=3D"noref=
errer" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sipcore</a><=
br>
<br>
_______________________________________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" rel=3D"noreferrer=
" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sipcore</a><br>
</blockquote></div><br></div>

--001a1141efced3d1d1053599f1a6--


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Subject: Re: [sipcore] Half outbound
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> On 19 Jun 2016, at 06:23, Ranjit Avasarala <ranjitkav0811@gmail.com> =
wrote:
>=20
> yes I agree - half outbound as a name does not sound appropriate - a =
more meaningful name is better
ABsolutely - but it was helpful in explaining what I wanted to do :-)

=E2=80=9CSIP Client Managed Connection Reuse=E2=80=9D may work better.

/O
>=20
> Regards
> Ranjit
>=20
> On Sat, Jun 18, 2016 at 2:45 PM, Olle E. Johansson <oej@edvina.net =
<mailto:oej@edvina.net>> wrote:
>=20
> > On 18 Jun 2016, at 11:02, Christer Holmberg =
<christer.holmberg@ericsson.com <mailto:christer.holmberg@ericsson.com>> =
wrote:
> >
> >
> > Hi,
> >
> > If you plan to bring this to DISPATCH, I suggest the following:
> >
> > 1. Don't call it "Half Outbound". It will people focus on the =
mechanism instead of what you actually want to achieve.
> >
> > 2. Write down a high-level use-case/call flow. Start by showing =
people what you want to do - not by showing what is wrong with existing =
mechanism (Outbound etc).
> >
> > 3. Based on the use-case/call-flow, write down a list of =
requirements. Again, leave the SIP protocol mechanism specifics out.
> >
> > 4. THEN you can start talking about protocol mechanisms. You can =
e.g. pick the parts of outbound that would fulfil the requirements. THAT =
you can call "half outbound".
> >
> > 5. If you are going to make statements like "people don't want to =
implement outbound", you really need some justification and background =
information. Otherwise people will just ask "how do you know people will =
implement this new mechanism?=E2=80=9D.
>=20
> Great feedback, thank you!
>=20
> /Olle
>=20
> >
> > Regards,
> >
> > Christer
> >
> >
> >
> > -----Original Message-----
> > From: sipcore [mailto:sipcore-bounces@ietf.org =
<mailto:sipcore-bounces@ietf.org>] On Behalf Of Olle E. Johansson
> > Sent: 17 June 2016 17:52
> > To: Paul Kyzivat <pkyzivat@alum.mit.edu =
<mailto:pkyzivat@alum.mit.edu>>
> > Cc: sipcore@ietf.org <mailto:sipcore@ietf.org>; Olle E Johansson =
<oej@edvina.net <mailto:oej@edvina.net>>
> > Subject: Re: [sipcore] Half outbound
> >
> >
> >> On 17 Jun 2016, at 16:48, Paul Kyzivat <pkyzivat@alum.mit.edu =
<mailto:pkyzivat@alum.mit.edu>> wrote:
> >>
> >> On 6/17/16 9:15 AM, Olle E. Johansson wrote:
> >>> http://www.slideshare.net/oej/sip-half-outbound-random-notes =
<http://www.slideshare.net/oej/sip-half-outbound-random-notes>
> >>>
> >>> Trying to summarize - am I right?
> >>>
> >>> - No one seems to oppose some work in this area
> >>> - No one seems to say =E2=80=9CYay=E2=80=9D for Outbound and stop =
doing alternatives
> >>>
> >>> As this is not an update to the core protocols, it seems like any
> >>> draft that can come out of this discussion needs to go to dispatch
> >>> and follow any route that group suggests.
> >>>
> >>> Anyone that wants to join me in writing a first draft to start the =
process?
> >>
> >> I thought we had concluded that half-outbound isn't needed, because =
the existing outbound may be used with a single connection/flow if =
properly configured.
> >>
> >> If people still don't want to do it, then maybe what we need is =
research into why.
> >
> > The SIPconnect team more or less refused to add outbound regardless =
on how many times I pointed at it as the only solution. I hope they can =
answer!
> >
> > So let=E2=80=99s take another round :-)
> >
> >
> > /O
> > _______________________________________________
> > 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>
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org <mailto:sipcore@ietf.org>
> https://www.ietf.org/mailman/listinfo/sipcore =
<https://www.ietf.org/mailman/listinfo/sipcore>
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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<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; -webkit-line-break: after-white-space;" =
class=3D""><br class=3D""><div><blockquote type=3D"cite" class=3D""><div =
class=3D"">On 19 Jun 2016, at 06:23, Ranjit Avasarala &lt;<a =
href=3D"mailto:ranjitkav0811@gmail.com" =
class=3D"">ranjitkav0811@gmail.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><div dir=3D"ltr" =
class=3D"">yes I agree - half outbound as a name does not sound =
appropriate - a more meaningful name is =
better</div></div></blockquote>ABsolutely - but it was helpful in =
explaining what I wanted to do :-)</div><div><br =
class=3D""></div><div>=E2=80=9CSIP Client Managed Connection Reuse=E2=80=9D=
 may work better.</div><div><br class=3D""></div><div>/O<br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D""><div =
dir=3D"ltr" class=3D""><div class=3D""><br class=3D""></div><div =
class=3D"">Regards</div><div class=3D"">Ranjit</div></div><div =
class=3D"gmail_extra"><br class=3D""><div class=3D"gmail_quote">On Sat, =
Jun 18, 2016 at 2:45 PM, Olle E. Johansson <span dir=3D"ltr" =
class=3D"">&lt;<a href=3D"mailto:oej@edvina.net" target=3D"_blank" =
class=3D"">oej@edvina.net</a>&gt;</span> wrote:<br class=3D""><blockquote =
class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex"><br class=3D"">
&gt; On 18 Jun 2016, at 11:02, Christer Holmberg &lt;<a =
href=3D"mailto:christer.holmberg@ericsson.com" =
class=3D"">christer.holmberg@ericsson.com</a>&gt; wrote:<br class=3D"">
&gt;<br class=3D"">
&gt;<br class=3D"">
&gt; Hi,<br class=3D"">
&gt;<br class=3D"">
&gt; If you plan to bring this to DISPATCH, I suggest the following:<br =
class=3D"">
&gt;<br class=3D"">
&gt; 1. Don't call it "Half Outbound". It will people focus on the =
mechanism instead of what you actually want to achieve.<br class=3D"">
&gt;<br class=3D"">
&gt; 2. Write down a high-level use-case/call flow. Start by showing =
people what you want to do - not by showing what is wrong with existing =
mechanism (Outbound etc).<br class=3D"">
&gt;<br class=3D"">
&gt; 3. Based on the use-case/call-flow, write down a list of =
requirements. Again, leave the SIP protocol mechanism specifics out.<br =
class=3D"">
&gt;<br class=3D"">
&gt; 4. THEN you can start talking about protocol mechanisms. You can =
e.g. pick the parts of outbound that would fulfil the requirements. THAT =
you can call "half outbound".<br class=3D"">
&gt;<br class=3D"">
&gt; 5. If you are going to make statements like "people don't want to =
implement outbound", you really need some justification and background =
information. Otherwise people will just ask "how do you know people will =
implement this new mechanism?=E2=80=9D.<br class=3D"">
<br class=3D"">
Great feedback, thank you!<br class=3D"">
<br class=3D"">
/Olle<br class=3D"">
<br class=3D"">
&gt;<br class=3D"">
&gt; Regards,<br class=3D"">
&gt;<br class=3D"">
&gt; Christer<br class=3D"">
&gt;<br class=3D"">
&gt;<br class=3D"">
&gt;<br class=3D"">
&gt; -----Original Message-----<br class=3D"">
&gt; From: sipcore [mailto:<a href=3D"mailto:sipcore-bounces@ietf.org" =
class=3D"">sipcore-bounces@ietf.org</a>] On Behalf Of Olle E. =
Johansson<br class=3D"">
&gt; Sent: 17 June 2016 17:52<br class=3D"">
&gt; To: Paul Kyzivat &lt;<a href=3D"mailto:pkyzivat@alum.mit.edu" =
class=3D"">pkyzivat@alum.mit.edu</a>&gt;<br class=3D"">
&gt; Cc: <a href=3D"mailto:sipcore@ietf.org" =
class=3D"">sipcore@ietf.org</a>; Olle E Johansson &lt;<a =
href=3D"mailto:oej@edvina.net" class=3D"">oej@edvina.net</a>&gt;<br =
class=3D"">
&gt; Subject: Re: [sipcore] Half outbound<br class=3D"">
&gt;<br class=3D"">
&gt;<br class=3D"">
&gt;&gt; On 17 Jun 2016, at 16:48, Paul Kyzivat &lt;<a =
href=3D"mailto:pkyzivat@alum.mit.edu" =
class=3D"">pkyzivat@alum.mit.edu</a>&gt; wrote:<br class=3D"">
&gt;&gt;<br class=3D"">
&gt;&gt; On 6/17/16 9:15 AM, Olle E. Johansson wrote:<br class=3D"">
&gt;&gt;&gt; <a =
href=3D"http://www.slideshare.net/oej/sip-half-outbound-random-notes" =
rel=3D"noreferrer" target=3D"_blank" =
class=3D"">http://www.slideshare.net/oej/sip-half-outbound-random-notes</a=
><br class=3D"">
&gt;&gt;&gt;<br class=3D"">
&gt;&gt;&gt; Trying to summarize - am I right?<br class=3D"">
&gt;&gt;&gt;<br class=3D"">
&gt;&gt;&gt; - No one seems to oppose some work in this area<br =
class=3D"">
&gt;&gt;&gt; - No one seems to say =E2=80=9CYay=E2=80=9D for Outbound =
and stop doing alternatives<br class=3D"">
&gt;&gt;&gt;<br class=3D"">
&gt;&gt;&gt; As this is not an update to the core protocols, it seems =
like any<br class=3D"">
&gt;&gt;&gt; draft that can come out of this discussion needs to go to =
dispatch<br class=3D"">
&gt;&gt;&gt; and follow any route that group suggests.<br class=3D"">
&gt;&gt;&gt;<br class=3D"">
&gt;&gt;&gt; Anyone that wants to join me in writing a first draft to =
start the process?<br class=3D"">
&gt;&gt;<br class=3D"">
&gt;&gt; I thought we had concluded that half-outbound isn't needed, =
because the existing outbound may be used with a single connection/flow =
if properly configured.<br class=3D"">
&gt;&gt;<br class=3D"">
&gt;&gt; If people still don't want to do it, then maybe what we need is =
research into why.<br class=3D"">
&gt;<br class=3D"">
&gt; The SIPconnect team more or less refused to add outbound regardless =
on how many times I pointed at it as the only solution. I hope they can =
answer!<br class=3D"">
&gt;<br class=3D"">
&gt; So let=E2=80=99s take another round :-)<br class=3D"">
&gt;<br class=3D"">
&gt;<br class=3D"">
&gt; /O<br class=3D"">
&gt; _______________________________________________<br class=3D"">
&gt; sipcore mailing list<br class=3D"">
&gt; <a href=3D"mailto:sipcore@ietf.org" =
class=3D"">sipcore@ietf.org</a><br class=3D"">
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" =
rel=3D"noreferrer" target=3D"_blank" =
class=3D"">https://www.ietf.org/mailman/listinfo/sipcore</a><br =
class=3D"">
&gt; _______________________________________________<br class=3D"">
&gt; sipcore mailing list<br class=3D"">
&gt; <a href=3D"mailto:sipcore@ietf.org" =
class=3D"">sipcore@ietf.org</a><br class=3D"">
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" =
rel=3D"noreferrer" target=3D"_blank" =
class=3D"">https://www.ietf.org/mailman/listinfo/sipcore</a><br =
class=3D"">
<br class=3D"">
_______________________________________________<br class=3D"">
sipcore mailing list<br class=3D"">
<a href=3D"mailto:sipcore@ietf.org" class=3D"">sipcore@ietf.org</a><br =
class=3D"">
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" =
rel=3D"noreferrer" target=3D"_blank" =
class=3D"">https://www.ietf.org/mailman/listinfo/sipcore</a><br =
class=3D"">
</blockquote></div><br class=3D""></div>
_______________________________________________<br class=3D"">sipcore =
mailing list<br class=3D""><a href=3D"mailto:sipcore@ietf.org" =
class=3D"">sipcore@ietf.org</a><br =
class=3D"">https://www.ietf.org/mailman/listinfo/sipcore<br =
class=3D""></div></blockquote></div><br class=3D""></body></html>=

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From: Roman Shpount <roman@telurix.com>
Date: Sun, 19 Jun 2016 21:28:02 -0400
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Subject: Re: [sipcore] SIPS must die - or at least be able to live without
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On Sat, Jun 18, 2016 at 8:47 PM, Asveren, Tolga <tasveren@sonusnet.com>
wrote:

> [TOLGA] I didn=E2=80=99t mean to imply that WebRTC+DTLS/SRTP solves any p=
roblem,
> to the contrary, it does not change anything fundamentally as far as the
> need for "end-to-end security/identity verification for signaling" is
> concerned.
> Use/trust on "fingerprint" does not necessarily imply "Identity
> verification", it merely serves the purpose of verifying that the
> certificate used for DTLS connection can be trusted on -as it is not sign=
ed
> by a CA-. And actually, if one takes things a bit more to philosophical
> level, isn=E2=80=99t even use of a CA "a transitive trust relationship"? =
How can I
> *really* trust a particular CA? They may (and actually it is a well-known
> fact that they are doing this in certain countries) issue certificates fo=
r
> Company-A to be used by some other entity. I think the only way I can tru=
st
> the identity on a certificate is if there is a 1:1 relationship between
> verified identity/corresponding CA. For example, Alex provides Bob his
> personal CA information (by whatever secure means, maybe handing it over =
in
> a diner) and Bob knows that this particular CA can be relied on only to
> verify identity=3DAlex.
> ZRTP would provide confidentiality but still would need end-to-end secure
> signaling from identity perspective.
> So, overall, the problem in front of us is a though nut to crack...
>

I just wanted to clarify what I've said earlier: There are two related but
separate problems:

1. End-to-end encryption. This means end to end protection of metadata

2. End-to-end authentication. This means that signaling information, such
as user metadata, destination addresses, ICE ufrags, and DTLS fingerprints
have not been modified.

Neither problem can be resolved without trusted third parties. If there is
a third party that both clients trust and with which both can authenticate
themselves, both problems can be resolved. There are some partial solutions
available without trusted third parties, but they can typically be
circumvented. In general, end-to-end authentication is a simpler problem to
solve, since all it requires is a third party, which both end points trust
that can generate a signature for the sensitive information, as well as a
mechanism to pass this signature through the signaling exchanges.

Regards,
_____________
Roman Shpount

--94eb2c05abe89ef4b90535ab9d6b
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure" data-smartmail=3D"gmail_signature"><br></div></div><div class=3D"gmail=
_quote">On Sat, Jun 18, 2016 at 8:47 PM, Asveren, Tolga <span dir=3D"ltr">&=
lt;<a href=3D"mailto:tasveren@sonusnet.com" target=3D"_blank">tasveren@sonu=
snet.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding=
-left:1ex">[TOLGA] I didn=E2=80=99t mean to imply that WebRTC+DTLS/SRTP sol=
ves any problem, to the contrary, it does not change anything fundamentally=
 as far as the need for &quot;end-to-end security/identity verification for=
 signaling&quot; is concerned.<br>
Use/trust on &quot;fingerprint&quot; does not necessarily imply &quot;Ident=
ity verification&quot;, it merely serves the purpose of verifying that the =
certificate used for DTLS connection can be trusted on -as it is not signed=
 by a CA-. And actually, if one takes things a bit more to philosophical le=
vel, isn=E2=80=99t even use of a CA &quot;a transitive trust relationship&q=
uot;? How can I *really* trust a particular CA? They may (and actually it i=
s a well-known fact that they are doing this in certain countries) issue ce=
rtificates for Company-A to be used by some other entity. I think the only =
way I can trust the identity on a certificate is if there is a 1:1 relation=
ship between verified identity/corresponding CA. For example, Alex provides=
 Bob his personal CA information (by whatever secure means, maybe handing i=
t over in a diner) and Bob knows that this particular CA can be relied on o=
nly to verify identity=3DAlex.<br>
ZRTP would provide confidentiality but still would need end-to-end secure s=
ignaling from identity perspective.<br>
So, overall, the problem in front of us is a though nut to crack...<br></bl=
ockquote><div><br></div><div>I just wanted to clarify what I&#39;ve said ea=
rlier: There are two related but separate problems:</div><div><br></div><di=
v>1. End-to-end encryption. This means end to end protection of metadata</d=
iv><div><br></div><div>2. End-to-end authentication. This means that signal=
ing information, such as user metadata, destination addresses, ICE ufrags, =
and DTLS fingerprints have not been modified.</div><div><br></div><div>Neit=
her problem can be resolved without trusted third parties. If there is a th=
ird party that both clients trust and with which both can authenticate them=
selves, both problems can be resolved. There are some partial solutions ava=
ilable without trusted third parties, but they can typically be circumvente=
d. In general, end-to-end authentication is a simpler problem to solve, sin=
ce all it requires is a third party, which both end points trust that can g=
enerate a signature for the sensitive information, as well as a mechanism t=
o pass this signature through the signaling exchanges.</div><div><br></div>=
<div>Regards,</div><div><div class=3D"gmail_signature" data-smartmail=3D"gm=
ail_signature">_____________<br>Roman Shpount</div></div><div>=C2=A0</div><=
/div></div></div>

--94eb2c05abe89ef4b90535ab9d6b--


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To: SIPCORE <sipcore@ietf.org>, "Paul Kyzivat" <pkyzivat@alum.mit.edu>, "SIPCORE Chairs" <sipcore-chairs@tools.ietf.org>
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Subject: Re: [sipcore] [Technical Errata Reported] RFC4596 (4716)
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(adding sipcore and an updated address for Paul)

Thoughts?

Thanks!

Ben.

On 20 Jun 2016, at 6:55, RFC Errata System wrote:

> The following errata report has been submitted for RFC4596,
> "Guidelines for Usage of the Session Initiation Protocol (SIP) Caller 
> Preferences Extension".
>
> --------------------------------------
> You may review the report below and at:
> http://www.rfc-editor.org/errata_search.php?rfc=4596&eid=4716
>
> --------------------------------------
> Type: Technical
> Reported by: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>
>
> Section: 3.9.2
>
> Original Text
> -------------
> languages=
>
> Corrected Text
> --------------
> language=
>
> Notes
> -----
> There are seven occurrences of this error in the sip examples of 
> section 3.9.2.
> As comparison, section 3.16.2 has the correct spelling in its three 
> occurrences.
> The parameter name refers to the language feature tag from RFC 2987, 
> named "language".
>
> Instructions:
> -------------
> This erratum is currently posted as "Reported". If necessary, please
> use "Reply All" to discuss whether it should be verified or
> rejected. When a decision is reached, the verifying party (IESG)
> can log in to change the status and edit the report, if necessary.
>
> --------------------------------------
> RFC4596 (draft-ietf-sipping-callerprefs-usecases-05)
> --------------------------------------
> Title               : Guidelines for Usage of the Session Initiation 
> Protocol (SIP) Caller Preferences Extension
> Publication Date    : July 2006
> Author(s)           : J. Rosenberg, P. Kyzivat
> Category            : INFORMATIONAL
> Source              : Session Initiation Proposal Investigation
> Area                : Real-time Applications and Infrastructure
> Stream              : IETF
> Verifying Party     : IESG


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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] [Technical Errata Reported] RFC4596 (4716)
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On 6/20/16 10:39 AM, Ben Campbell wrote:
> (adding sipcore and an updated address for Paul)
>
> Thoughts?

This seems like a real error. The name of the feature tag registered in 
the IANA feature tag registry is "language".

	Thanks,
	Paul

> Thanks!
>
> Ben.
>
> On 20 Jun 2016, at 6:55, RFC Errata System wrote:
>
>> The following errata report has been submitted for RFC4596,
>> "Guidelines for Usage of the Session Initiation Protocol (SIP) Caller
>> Preferences Extension".
>>
>> --------------------------------------
>> You may review the report below and at:
>> http://www.rfc-editor.org/errata_search.php?rfc=4596&eid=4716
>>
>> --------------------------------------
>> Type: Technical
>> Reported by: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>
>>
>> Section: 3.9.2
>>
>> Original Text
>> -------------
>> languages=
>>
>> Corrected Text
>> --------------
>> language=
>>
>> Notes
>> -----
>> There are seven occurrences of this error in the sip examples of
>> section 3.9.2.
>> As comparison, section 3.16.2 has the correct spelling in its three
>> occurrences.
>> The parameter name refers to the language feature tag from RFC 2987,
>> named "language".
>>
>> Instructions:
>> -------------
>> This erratum is currently posted as "Reported". If necessary, please
>> use "Reply All" to discuss whether it should be verified or
>> rejected. When a decision is reached, the verifying party (IESG)
>> can log in to change the status and edit the report, if necessary.
>>
>> --------------------------------------
>> RFC4596 (draft-ietf-sipping-callerprefs-usecases-05)
>> --------------------------------------
>> Title               : Guidelines for Usage of the Session Initiation
>> Protocol (SIP) Caller Preferences Extension
>> Publication Date    : July 2006
>> Author(s)           : J. Rosenberg, P. Kyzivat
>> Category            : INFORMATIONAL
>> Source              : Session Initiation Proposal Investigation
>> Area                : Real-time Applications and Infrastructure
>> Stream              : IETF
>> Verifying Party     : IESG
>


From nobody Mon Jun 20 08:47:52 2016
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To: Paul Kyzivat <pkyzivat@alum.mit.edu>, Ben Campbell <ben@nostrum.com>, SIPCORE <sipcore@ietf.org>, SIPCORE Chairs <sipcore-chairs@tools.ietf.org>
Thread-Topic: [sipcore] [Technical Errata Reported] RFC4596 (4716)
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Subject: Re: [sipcore] [Technical Errata Reported] RFC4596 (4716)
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+1

Regards,

Christer

Sent from my Windows Phone
________________________________
From: Paul Kyzivat<mailto:pkyzivat@alum.mit.edu>
Sent: =FD20/=FD06/=FD2016 18:30
To: Ben Campbell<mailto:ben@nostrum.com>; SIPCORE<mailto:sipcore@ietf.org>;=
 SIPCORE Chairs<mailto:sipcore-chairs@tools.ietf.org>
Cc: jdrosen@cisco.com<mailto:jdrosen@cisco.com>; aamelnikov@fastmail.fm<mai=
lto:aamelnikov@fastmail.fm>; gunnar.hellstrom@omnitor.se<mailto:gunnar.hell=
strom@omnitor.se>; pkyzivat@cisco.com<mailto:pkyzivat@cisco.com>; alissa@co=
operw.in<mailto:alissa@cooperw.in>; Gonzalo Camarillo<mailto:gonzalo.camari=
llo@ericsson.com>
Subject: Re: [sipcore] [Technical Errata Reported] RFC4596 (4716)

On 6/20/16 10:39 AM, Ben Campbell wrote:
> (adding sipcore and an updated address for Paul)
>
> Thoughts?

This seems like a real error. The name of the feature tag registered in
the IANA feature tag registry is "language".

        Thanks,
        Paul

> Thanks!
>
> Ben.
>
> On 20 Jun 2016, at 6:55, RFC Errata System wrote:
>
>> The following errata report has been submitted for RFC4596,
>> "Guidelines for Usage of the Session Initiation Protocol (SIP) Caller
>> Preferences Extension".
>>
>> --------------------------------------
>> You may review the report below and at:
>> http://www.rfc-editor.org/errata_search.php?rfc=3D4596&eid=3D4716
>>
>> --------------------------------------
>> Type: Technical
>> Reported by: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>
>>
>> Section: 3.9.2
>>
>> Original Text
>> -------------
>> languages=3D
>>
>> Corrected Text
>> --------------
>> language=3D
>>
>> Notes
>> -----
>> There are seven occurrences of this error in the sip examples of
>> section 3.9.2.
>> As comparison, section 3.16.2 has the correct spelling in its three
>> occurrences.
>> The parameter name refers to the language feature tag from RFC 2987,
>> named "language".
>>
>> Instructions:
>> -------------
>> This erratum is currently posted as "Reported". If necessary, please
>> use "Reply All" to discuss whether it should be verified or
>> rejected. When a decision is reached, the verifying party (IESG)
>> can log in to change the status and edit the report, if necessary.
>>
>> --------------------------------------
>> RFC4596 (draft-ietf-sipping-callerprefs-usecases-05)
>> --------------------------------------
>> Title               : Guidelines for Usage of the Session Initiation
>> Protocol (SIP) Caller Preferences Extension
>> Publication Date    : July 2006
>> Author(s)           : J. Rosenberg, P. Kyzivat
>> Category            : INFORMATIONAL
>> Source              : Session Initiation Proposal Investigation
>> Area                : Real-time Applications and Infrastructure
>> Stream              : IETF
>> Verifying Party     : IESG
>

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

--_000_7594FB04B1934943A5C02806D1A2204B380EE3AEESESSMB209erics_
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ding-left: 4pt; border-left: #800000 2px solid; } --></style>
</head>
<body>
<div>
<div>
<div style=3D"font-family:Calibri,sans-serif; font-size:11pt">&#43;1<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">=FD20=
/=FD06/=FD2016 18:30</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:ben@nostrum.com">Ben Campbell</a>;
<a href=3D"mailto:sipcore@ietf.org">SIPCORE</a>; <a href=3D"mailto:sipcore-=
chairs@tools.ietf.org">
SIPCORE Chairs</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Cc:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:jdrosen@cisco.com">jdrosen@cisco.com</a>;
<a href=3D"mailto:aamelnikov@fastmail.fm">aamelnikov@fastmail.fm</a>; <a hr=
ef=3D"mailto:gunnar.hellstrom@omnitor.se">
gunnar.hellstrom@omnitor.se</a>; <a href=3D"mailto:pkyzivat@cisco.com">pkyz=
ivat@cisco.com</a>;
<a href=3D"mailto:alissa@cooperw.in">alissa@cooperw.in</a>; <a href=3D"mail=
to:gonzalo.camarillo@ericsson.com">
Gonzalo Camarillo</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] [Technical Errata Reported] RFC4596 (4716)</span><br>
<br>
</div>
</div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">On 6/20/16 10:39 AM, Ben Campbell wrote:<br>
&gt; (adding sipcore and an updated address for Paul)<br>
&gt;<br>
&gt; Thoughts?<br>
<br>
This seems like a real error. The name of the feature tag registered in <br=
>
the IANA feature tag registry is &quot;language&quot;.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Thanks,<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paul<br>
<br>
&gt; Thanks!<br>
&gt;<br>
&gt; Ben.<br>
&gt;<br>
&gt; On 20 Jun 2016, at 6:55, RFC Errata System wrote:<br>
&gt;<br>
&gt;&gt; The following errata report has been submitted for RFC4596,<br>
&gt;&gt; &quot;Guidelines for Usage of the Session Initiation Protocol (SIP=
) Caller<br>
&gt;&gt; Preferences Extension&quot;.<br>
&gt;&gt;<br>
&gt;&gt; --------------------------------------<br>
&gt;&gt; You may review the report below and at:<br>
&gt;&gt; <a href=3D"http://www.rfc-editor.org/errata_search.php?rfc=3D4596&=
amp;eid=3D4716">http://www.rfc-editor.org/errata_search.php?rfc=3D4596&amp;=
eid=3D4716</a><br>
&gt;&gt;<br>
&gt;&gt; --------------------------------------<br>
&gt;&gt; Type: Technical<br>
&gt;&gt; Reported by: Gunnar Hellstrom &lt;gunnar.hellstrom@omnitor.se&gt;<=
br>
&gt;&gt;<br>
&gt;&gt; Section: 3.9.2<br>
&gt;&gt;<br>
&gt;&gt; Original Text<br>
&gt;&gt; -------------<br>
&gt;&gt; languages=3D<br>
&gt;&gt;<br>
&gt;&gt; Corrected Text<br>
&gt;&gt; --------------<br>
&gt;&gt; language=3D<br>
&gt;&gt;<br>
&gt;&gt; Notes<br>
&gt;&gt; -----<br>
&gt;&gt; There are seven occurrences of this error in the sip examples of<b=
r>
&gt;&gt; section 3.9.2.<br>
&gt;&gt; As comparison, section 3.16.2 has the correct spelling in its thre=
e<br>
&gt;&gt; occurrences.<br>
&gt;&gt; The parameter name refers to the language feature tag from RFC 298=
7,<br>
&gt;&gt; named &quot;language&quot;.<br>
&gt;&gt;<br>
&gt;&gt; Instructions:<br>
&gt;&gt; -------------<br>
&gt;&gt; This erratum is currently posted as &quot;Reported&quot;. If neces=
sary, please<br>
&gt;&gt; use &quot;Reply All&quot; to discuss whether it should be verified=
 or<br>
&gt;&gt; rejected. When a decision is reached, the verifying party (IESG)<b=
r>
&gt;&gt; can log in to change the status and edit the report, if necessary.=
<br>
&gt;&gt;<br>
&gt;&gt; --------------------------------------<br>
&gt;&gt; RFC4596 (draft-ietf-sipping-callerprefs-usecases-05)<br>
&gt;&gt; --------------------------------------<br>
&gt;&gt; Title&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; : Guidelines for Usage of the Session Initiation<br=
>
&gt;&gt; Protocol (SIP) Caller Preferences Extension<br>
&gt;&gt; Publication Date&nbsp;&nbsp;&nbsp; : July 2006<br>
&gt;&gt; Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp; : J. Rosenberg, P. Kyzivat<br>
&gt;&gt; Category&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; : INFORMATIONAL<br>
&gt;&gt; Source&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp; : Session Initiation Proposal Investigation<br>
&gt;&gt; Area&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp; : Real-time Applications and Infrastructure<br=
>
&gt;&gt; Stream&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp; : IETF<br>
&gt;&gt; Verifying Party&nbsp;&nbsp;&nbsp;&nbsp; : IESG<br>
&gt;<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>
</div>
</span></font>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B380EE3AEESESSMB209erics_--


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From nobody Mon Jun 20 09:56:06 2016
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From: "Ben Campbell" <ben@nostrum.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>
Date: Mon, 20 Jun 2016 11:21:07 -0500
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Cc: SIPCORE Chairs <sipcore-chairs@tools.ietf.org>, "jdrosen@cisco.com" <jdrosen@cisco.com>, "aamelnikov@fastmail.fm" <aamelnikov@fastmail.fm>, "alissa@cooperw.in" <alissa@cooperw.in>, "pkyzivat@cisco.com" <pkyzivat@cisco.com>, SIPCORE <sipcore@ietf.org>, "gunnar.hellstrom@omnitor.se" <gunnar.hellstrom@omnitor.se>, Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>
Subject: Re: [sipcore] [Technical Errata Reported] RFC4596 (4716)
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+1 from me, too. I plan to approve it.

Ben.

On 20 Jun 2016, at 10:37, Christer Holmberg wrote:

> +1
>
> Regards,
>
> Christer
>
> Sent from my Windows Phone
> ________________________________
> From: Paul Kyzivat<mailto:pkyzivat@alum.mit.edu>
> Sent: â€Ž20/â€Ž06/â€Ž2016 18:30
> To: Ben Campbell<mailto:ben@nostrum.com>; 
> SIPCORE<mailto:sipcore@ietf.org>; SIPCORE 
> Chairs<mailto:sipcore-chairs@tools.ietf.org>
> Cc: jdrosen@cisco.com<mailto:jdrosen@cisco.com>; 
> aamelnikov@fastmail.fm<mailto:aamelnikov@fastmail.fm>; 
> gunnar.hellstrom@omnitor.se<mailto:gunnar.hellstrom@omnitor.se>; 
> pkyzivat@cisco.com<mailto:pkyzivat@cisco.com>; 
> alissa@cooperw.in<mailto:alissa@cooperw.in>; Gonzalo 
> Camarillo<mailto:gonzalo.camarillo@ericsson.com>
> Subject: Re: [sipcore] [Technical Errata Reported] RFC4596 (4716)
>
> On 6/20/16 10:39 AM, Ben Campbell wrote:
>> (adding sipcore and an updated address for Paul)
>>
>> Thoughts?
>
> This seems like a real error. The name of the feature tag registered 
> in
> the IANA feature tag registry is "language".
>
>         Thanks,
>         Paul
>
>> Thanks!
>>
>> Ben.
>>
>> On 20 Jun 2016, at 6:55, RFC Errata System wrote:
>>
>>> The following errata report has been submitted for RFC4596,
>>> "Guidelines for Usage of the Session Initiation Protocol (SIP) 
>>> Caller
>>> Preferences Extension".
>>>
>>> --------------------------------------
>>> You may review the report below and at:
>>> http://www.rfc-editor.org/errata_search.php?rfc=4596&eid=4716
>>>
>>> --------------------------------------
>>> Type: Technical
>>> Reported by: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>
>>>
>>> Section: 3.9.2
>>>
>>> Original Text
>>> -------------
>>> languages=
>>>
>>> Corrected Text
>>> --------------
>>> language=
>>>
>>> Notes
>>> -----
>>> There are seven occurrences of this error in the sip examples of
>>> section 3.9.2.
>>> As comparison, section 3.16.2 has the correct spelling in its three
>>> occurrences.
>>> The parameter name refers to the language feature tag from RFC 2987,
>>> named "language".
>>>
>>> Instructions:
>>> -------------
>>> This erratum is currently posted as "Reported". If necessary, please
>>> use "Reply All" to discuss whether it should be verified or
>>> rejected. When a decision is reached, the verifying party (IESG)
>>> can log in to change the status and edit the report, if necessary.
>>>
>>> --------------------------------------
>>> RFC4596 (draft-ietf-sipping-callerprefs-usecases-05)
>>> --------------------------------------
>>> Title               : Guidelines for Usage of the Session Initiation
>>> Protocol (SIP) Caller Preferences Extension
>>> Publication Date    : July 2006
>>> Author(s)           : J. Rosenberg, P. Kyzivat
>>> Category            : INFORMATIONAL
>>> Source              : Session Initiation Proposal Investigation
>>> Area                : Real-time Applications and Infrastructure
>>> Stream              : IETF
>>> Verifying Party     : IESG
>>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From nobody Mon Jun 20 12:26:23 2016
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From: "Ben Campbell" <ben@nostrum.com>
To: draft-holmberg-dispatch-rfc7315-updates-06.all@ietf.org
Date: Mon, 20 Jun 2016 14:26:18 -0500
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Subject: [sipcore] AD Evaluation of draft-holmberg-dispatch-rfc7315-updates-06
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(copying sipcore since, even though this is not a wg draft, sipcore 
probably offers the best nexus of expertise.)

Hi,

This is my AD Evaluation of draft-holmberg-dispatch-rfc7315-updates-06. 
I have one substantive comment and one editorial comment. I don't think 
either of these need block IETF last call. They can be handled along 
with any other last call feedback.

Thanks!

Ben.

--------------

Substantive:

The security considerations state that the draft removes some places 
that some of the P-Headers can be sent, but expands that to some other 
places. Further, it says that neither introduce new security 
considerations beyond those in 7315.

I accept that for the reduction part. But I'm not sure we can state that 
sort of thing for the expansion part, at least without some more 
discussion. Since 7315 already acknowledges potential privacy issues 
around P-Access-Network-Info, I'd like to at least see a sentence or two 
about the allowance of that in ACK requests, even if they just say that 
this addition makes things no worse than they already are.

Editorial:

-5, first sentence: "The security considerations for P- header fields 
are defined in
    [RFC7315]"
I assume this means 7315 discusses the security considerations for these 
P-Headers specifically, not P-Headers in general. Is this the intent? If 
so, I suggest:

s/... for P-header fields.../ ... for these P-header fields...


From nobody Mon Jun 20 12:27:32 2016
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From: "Ben Campbell" <ben@nostrum.com>
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(copying sipcore since, even though this is not a wg draft, sipcore 
probably offers the best nexus of expertise.)

Hi,

This is my AD Evaluation of draft-holmberg-dispatch-rfc7315-updates-06. 
I have one substantive comment and one editorial comment. I don't think 
either of these need block IETF last call. They can be handled along 
with any other last call feedback.

Thanks!

Ben.

--------------

Substantive:

The security considerations state that the draft removes some places 
that some of the P-Headers can be sent, but expands that to some other 
places. Further, it says that neither introduce new security 
considerations beyond those in 7315.

I accept that for the reduction part. But I'm not sure we can state that 
sort of thing for the expansion part, at least without some more 
discussion. Since 7315 already acknowledges potential privacy issues 
around P-Access-Network-Info, I'd like to at least see a sentence or two 
about the allowance of that in ACK requests, even if they just say that 
this addition makes things no worse than they already are.

Editorial:

-5, first sentence: "The security considerations for P- header fields 
are defined in
    [RFC7315]"
I assume this means 7315 discusses the security considerations for these 
P-Headers specifically, not P-Headers in general. Is this the intent? If 
so, I suggest:

s/... for P-header fields.../ ... for these P-header fields...


From nobody Mon Jun 20 14:16:09 2016
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From: "Ben Campbell" <ben@nostrum.com>
To: draft-ietf-sipcore-dns-dual-stack.all@ietf.org, SIPCORE <sipcore@ietf.org>
Date: Mon, 20 Jun 2016 16:16:01 -0500
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Hi,

This is my AD evaluation of draft-ietf-sipcore-dns-dual-stack-06. I only 
have a few editorial comments. None of these need to delay IETF last 
call, and can be handled along with any feedback from the last call.

Thanks!

Ben.

---------

-1, 2nd paragraph, first sentence:

That sentence is a bit hard to parse, please consider breaking into two 
or more simpler sentences.

-2, definition of IPv4/IPv6 UA/UAC/UAS:

This defines one term, then declares the draft will use another. Why not 
just define _that_ term in the first place?

-4:

This section kind of buries the lede, and leaves me wondering what the 
actual clarification was. Is the point to say not to interleave the 
results of the address record queries among srv records? Is there a 
concern that people thought they were to do something different? It 
would be helpful to say what misconception you intend to clarify.

-5, first paragraph, last sentence:

Oddly constructed sentence. The structure seems to imply a tension 
between the idea of optimizing performance and not introducing security 
considerations. Please consider something to the effect of the 
following:

â€œBoth of these procedures are optimization designed to improve the 
performance of dual-stack clients. Neither introduce new security 
considerations.â€


From nobody Mon Jun 20 14:29:03 2016
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From: worley@ariadne.com (Dale R. Worley)
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Subject: [sipcore] Happy Eyeballs: Requirements
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Getting back involved here:

I've gone over the discussions, and I think I can extract a good
statement of the requirements we need to satisfy for Happy Eyeballs in
practice.  Perhaps some of these can be extracted as seperate work
items, but they all are closely related.  Some of them can be solved
fairly simply, but I think some will involve significant new work.  The
major item that is outside its scope is signaling privacy.

Needless to say, copious comments are invited.  I'll send further
messages outlining what I think are the state of solutions for various
items.

- Solve the "happy eyeballs" problem for INVITE, i.e., if a
  destination URI is advertised with both multiple targets (IPv4 and
  IPv6 or otherwise), but connectivity to only one target exists,
  usually message transmission will not be delayed unacceptably, and in
  particular, not by a full timeout as prescribed in 3261.

Note that the solution may be decidedly different based on the
transport protocol(s) for the targets.

There is uncertainty about whether adding an RTT to signaling times is
acceptable.  For example, in practice RTT on the overall Internet is
less than 1/2 second, and that is an acceptable delay in call setup
times under ordinary circumstances.

- The solution must approximate 3263, in that if several targets are
  specified by *different* SRV records and are reachable over a long
  period of time, the relative traffic shares sent to them must be
  compatible with the priorities and weights of the SRV records.  (If
  they are alternatives that are *not* derived from different SRV
  records, the solution is unconstrained regarding relative traffic
  shares.)

- The solution should be as compatible with SIP Outbound as practicable.

The difficult part of this is "the solution for simple UDP (i.e., not
using SIP Outbound)".  There is a strong perception that in systems
where an edge proxy services 10^5 or 10^6 UAs that only
UDP-without-Outbound has a low enough per-flow overhead to be
workable.  The important requirements for simple UDP seem to be:

- The per-UA state in the edge proxy must be no larger than about what
  is kept for a registration.

- The update rate of the per-UA state in the edge proxy must be no
  larger than about the update rate for a registration.

It is perceived that TCP, TLS, and even UDP-with-Outbound do not meet
these requirements.  A substantial fraction of the UAs tested in the
latest SIPit do not support TCP, TLS, or Outbound, showing that there
is substantial market presence of simple-UDP-only UAs.

It is possible that a "Simple Outbound" can be defined that provides
the needed part of the functionality of Outbound at a sufficiently low
overhead and complexity that it meets these requirements.  If so, it
is desirable that it be conceptually and operationally
upward-compatible with Outbound.

- Deal with the handoff problem, i.e., when the call (signaling and
  media) are being sent over one interface, and that interface becomes
  unavailable, but another interface has been available, the signaling
  and media should be rerouted via the other interface.

  - The primary requirement is that the signaling path is
    reestablished, i.e., the call is not dropped.  It may take several
    seconds for signaling flow to be reestablished.

  - The media flow should be interrupted for only a "short" period of
    time so the user does not assume that the call has been dropped.
    2 seconds seems a reasonable target.

- Clients behind client-side NATs must be fully supported.

- In particular, if the external address of a NAT binding changes, the
  UA must be able to recover, with the same requirements as an
  interface handoff.

- Signaling privacy (e.g., "SIP over DTLS") is not in scope.

Dale


From nobody Mon Jun 20 14:42:33 2016
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Subject: [sipcore] Last Call: <draft-ietf-sipcore-dns-dual-stack-06.txt> (Locating Session Initiation Protocol (SIP) Servers in a Dual-Stack IP Network) to Proposed Standard
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The IESG has received a request from the Session Initiation Protocol Core
WG (sipcore) to consider the following document:
- 'Locating Session Initiation Protocol (SIP) Servers in a Dual-Stack IP
   Network'
  <draft-ietf-sipcore-dns-dual-stack-06.txt> as Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action. Please send substantive comments to the
ietf@ietf.org mailing lists by 2016-07-04. Exceptionally, comments may be
sent to iesg@ietf.org instead. In either case, please retain the
beginning of the Subject line to allow automated sorting.

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 updates the DNS procedures
   described in RFC 3263 for dual-stack SIP implementations in
   preparation for forthcoming specifications for applying Happy
   Eyeballs principles to SIP.




The file can be obtained via
https://datatracker.ietf.org/doc/draft-ietf-sipcore-dns-dual-stack/

IESG discussion can be tracked via
https://datatracker.ietf.org/doc/draft-ietf-sipcore-dns-dual-stack/ballot/


No IPR declarations have been submitted directly on this I-D.



From nobody Tue Jun 21 01:27:05 2016
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Ben Campbell <ben@nostrum.com>, "draft-holmberg-dispatch-rfc7315-updates.all@ietf.org" <draft-holmberg-dispatch-rfc7315-updates.all@ietf.org>
Thread-Topic: AD Evaluation of draft-holmberg-dispatch-rfc7315-updates-06
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Date: Tue, 21 Jun 2016 08:26:59 +0000
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Subject: Re: [sipcore] AD Evaluation of draft-holmberg-dispatch-rfc7315-updates-06
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Hi Ben,

See inline.

>--------------
>
>Substantive:
>
>The security considerations state that the draft removes some places
>that some of the P-Headers can be sent, but expands that to some other
>places. Further, it says that neither introduce new security
>considerations beyond those in 7315.
>
>I accept that for the reduction part. But I'm not sure we can state that
>sort of thing for the expansion part, at least without some more
>discussion. Since 7315 already acknowledges potential privacy issues
>around P-Access-Network-Info, I'd like to at least see a sentence or two
>about the allowance of that in ACK requests, even if they just say that
>this addition makes things no worse than they already are.


OLD:

The security considerations for P- header fields are defined in
   [RFC7315].  This specification allows some header fields to be
   present in messages where they were previously not allowed, and
   disallow some header fields to be present in messages where they were
   previously allowed. That does not cause any security issues, but
   implementations need to be aware that implementations may not have
   been updated according to this document, and take proper actions if a
   header field occur, or does not occur, in a message where it should
   occur (or occurs in a message where it should not occur).



NEW:

The security considerations for these P- header fields are defined in
   [RFC7315].  This specification allows some header fields to be
   present in messages where they were previously not allowed, and the
security considerations and assumptions (e.g. regarding only sending
Information to trusted entities) also to those messages. In addition,
this specification also disallow some header fields to be present
in message where they were previously allowed. That does not cause
any security issues, but implementations need to be aware that
implementations may not have been updated according to this document,
and take proper actions if a header field occur, or does not occur,
in a message where it should occur (or occurs in a message where it
should not occur).



>Editorial:
>
>-5, first sentence: "The security considerations for P- header fields
>are defined in
>    [RFC7315]"
>I assume this means 7315 discusses the security considerations for these
>P-Headers specifically, not P-Headers in general. Is this the intent? If
>so, I suggest:
>
>s/... for P-header fields.../ ... for these P-header fields...

I=B9ll fix as suggested (ass new text above).

Regards,

Christer


From nobody Tue Jun 21 08:22:31 2016
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From: "Ben Campbell" <ben@nostrum.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>
Date: Tue, 21 Jun 2016 10:22:24 -0500
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That's a good start, but don't be surprised if we get questions 
specifically about adding NPLI to ACK requests. some language to the 
effect of the following might help:

"This document adds the ability to include P-Access-Network-Info in ACK 
requests. As documented in RFC7315, P-Access-Network-Info may include 
privacy sensitive information, including the user's location. The 
security and privacy considerations for P-Access-Network-Info in ACK 
requests are similar to those for the other SIP requests discussed in 
RFC7315."

Thanks!

Ben.

On 21 Jun 2016, at 3:26, Christer Holmberg wrote:

> Hi Ben,
>
> See inline.
>
>> --------------
>>
>> Substantive:
>>
>> The security considerations state that the draft removes some places
>> that some of the P-Headers can be sent, but expands that to some 
>> other
>> places. Further, it says that neither introduce new security
>> considerations beyond those in 7315.
>>
>> I accept that for the reduction part. But I'm not sure we can state 
>> that
>> sort of thing for the expansion part, at least without some more
>> discussion. Since 7315 already acknowledges potential privacy issues
>> around P-Access-Network-Info, I'd like to at least see a sentence or 
>> two
>> about the allowance of that in ACK requests, even if they just say 
>> that
>> this addition makes things no worse than they already are.
>
>
> OLD:
>
> The security considerations for P- header fields are defined in
>    [RFC7315].  This specification allows some header fields to be
>    present in messages where they were previously not allowed, and
>    disallow some header fields to be present in messages where they 
> were
>    previously allowed. That does not cause any security issues, but
>    implementations need to be aware that implementations may not have
>    been updated according to this document, and take proper actions if 
> a
>    header field occur, or does not occur, in a message where it should
>    occur (or occurs in a message where it should not occur).
>
>
>
> NEW:
>
> The security considerations for these P- header fields are defined in
>    [RFC7315].  This specification allows some header fields to be
>    present in messages where they were previously not allowed, and the
> security considerations and assumptions (e.g. regarding only sending
> Information to trusted entities) also to those messages. In addition,
> this specification also disallow some header fields to be present
> in message where they were previously allowed. That does not cause
> any security issues, but implementations need to be aware that
> implementations may not have been updated according to this document,
> and take proper actions if a header field occur, or does not occur,
> in a message where it should occur (or occurs in a message where it
> should not occur).
>
>
>
>> Editorial:
>>
>> -5, first sentence: "The security considerations for P- header fields
>> are defined in
>>    [RFC7315]"
>> I assume this means 7315 discusses the security considerations for 
>> these
>> P-Headers specifically, not P-Headers in general. Is this the intent? 
>> If
>> so, I suggest:
>>
>> s/... for P-header fields.../ ... for these P-header fields...
>
> IÂ¹ll fix as suggested (ass new text above).
>
> Regards,
>
> Christer


From nobody Tue Jun 21 09:53:02 2016
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From: "Gonzalo Salgueiro (gsalguei)" <gsalguei@cisco.com>
To: Ben Campbell <ben@nostrum.com>
Thread-Topic: AD Evaluation of draft-holmberg-dispatch-rfc7315-updates-06
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Thread-Topic: [sipcore] SIP AuthNZ Problem Statement - v3
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Date: Tue, 21 Jun 2016 23:46:10 +0000
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Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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--_000_D38F19C319F959jonpetersonneustarbiz_
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It's not really what I'm looking for, no, and after mulling this over a bit=
, I fear going farther along this path would only drill down into levels of=
 complexity that will likely make this harder to understand rather than eas=
ier.

I do acknowledge that you describe the INVITE as being rejected if it doesn=
't contain a token - but that's not the same thing as saying the INVITE wil=
l be rejected because the relying party makes an authorization decision bas=
ed on the attributes in the token. It's more like an INVITE being rejected =
because a user agent can't supply an Authorization header. I will reiterate=
 that I don't think the kinds of attributes you are proposing should be in =
the scope of an effort that will try to address a framework for authorizing=
 SIP requests. I don't think the values of the attributes (as opposed to th=
e mere presence of some attributes) have anything to do with authorizing th=
e SIP request that you propose would carry the attributes.

With regards to this whole conferencing example, it does seem a bit contriv=
ed to me, like you're sketching a SIP-based way of setting conference polic=
y on the fly that happens to require the kind of attributes you want, and t=
hen saying that demonstrates the need for the attributes. The fact that all=
 of the ways to do conference policy we can point to in fact do it via the =
web makes it hard for me to understand why we should create a SIP-based way=
 of doing this. To me, it looks like you're opening a problem space that ha=
s much more to do with reshaping the way that SIP does conferences (and aga=
in, as Radhika's mail pointing out, there's been a lot of thinking about th=
is since the days of RFC4579) than it does with defining a framework for au=
thorizing SIP requests.

Jon Peterson
Neustar, Inc.

From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com<mailto:rifaat.ietf@gmail.co=
m>>
Date: Thursday, June 16, 2016 at 6:19 AM
To: Jon Peterson <jon.peterson@neustar.biz<mailto:jon.peterson@neustar.biz>=
>
Cc: "sipcore@ietf.org<mailto:sipcore@ietf.org>" <sipcore@ietf.org<mailto:si=
pcore@ietf.org>>
Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3

The attributes I provided are examples of attributes that might be provided=
 in an access token; it was not meant to include all the attributes for thi=
s use case, because my intention was not to focus on conference server, but=
 to discuss the idea in general.

The following could be used with the conference factory to request a confer=
ence focus:
The INVITE sent to the conference factory could include a token that has a =
conference-type attribute which could take one of the following values: aud=
io, video, webconference, audio-webconference, or video-webconference.
The conference factory will then issue a conference uri for the requested c=
onference type, assuming the token is valid. If the conference-type attribu=
te is missing, then the request would be rejected.

Is that what you are looking for?

Regards,
 Rifaat





On Wed, Jun 15, 2016 at 5:06 PM, Peterson, Jon <jon.peterson@neustar.biz<ma=
ilto:jon.peterson@neustar.biz>> wrote:


In that case, if you could furnish a use case with attributes that would be=
 carried in a SIP iNVITE that are actually salient to the authorization of =
that particular SIP request, we'd be getting a lot closer to common ground.


An obvious example would be an INVITE with a token sent to a conference fac=
tory to authorize the creation of a focus.

We're still talking past each other. My point above was that the two attrib=
utes you specified - constraints on conference size and media types - have =
nothing to do with the authorization of the INVITE carrying those attribute=
s. Like, unless the conference size permitted is 0, then the INVITE that cr=
eates an ad hoc conference and connects you to its bridge is not going to b=
e rejected by the focus on the basis of that conference size attribute. If =
the maximum conference size were 0, the AuthZ service just wouldn't attach =
a token to an INVITE as it would never be allowed to start a conference. Sa=
me goes for media types.

Instead, what your two attributes constrain is what happens after the confe=
rence is created: when ten users try to dial in to a conference with a perm=
itted size of nine, say. That tenth INVITE would get rejected - not the con=
ference-forming INVITE that you propose would carry the token. That's why w=
hat you're proposing is overloading the INVITE to tunnel conference policy =
constraints to the focus. I mean, maybe that's a problem space worth invest=
igating, though I suspect there are good reasons why RFC4353 specified that=
 conference policy is not manipulated with SIP - mostly because the semanti=
cs of manipulating conference policy probably don't map well onto dialog-fo=
rming SIP requests. But I think that conference policy provisioning problem=
 space would be pretty far from where we started in this discussion.

Let me reiterate what I wrote in my previous mail, which I quote here at th=
e top: "furnish a use case with attributes that would be carried in a SIP i=
NVITE that are actually salient to the authorization of that particular SIP=
 request." Again: "that particular SIP request." Not later transactions in =
some broader context in which the original user agent probably isn't even a=
n endpoint.

The Authorization header in SIP, for example, is actually used by the UAS t=
o make a decision about whether or not to authorize the request in which th=
e header appears.  The token that you propose a SIP INVITE would carry in t=
his case is not used by the relying party to make a decision about whether =
or not that INVITE is authorized. That's why I believe this conference poli=
cy issue is a different problem space entirely from figuring out how to imp=
lement attribute-based authorization for SIP requests.

Jon Peterson
Neustar, Inc.


Regards,
 Rifaat



--_000_D38F19C319F959jonpetersonneustarbiz_
Content-Type: text/html; charset="us-ascii"
Content-ID: <DEAB1FBE04D6DA448665FE06208CF5BA@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>It's not really what I'm looking for, no, and after mulling this over =
a bit, I fear going farther along this path would only drill down into leve=
ls of complexity that will likely make this harder to understand rather tha=
n easier.</div>
<div><br>
</div>
<div>I do acknowledge that you describe the INVITE as being rejected if it =
doesn't contain a token - but that's not the same thing as saying the INVIT=
E will be rejected because the relying party makes an authorization decisio=
n based on the attributes in the
 token. It's more like an INVITE being rejected because a user agent can't =
supply an Authorization header. I will reiterate that I don't think the kin=
ds of attributes you are proposing should be in the scope of an effort that=
 will try to address a framework
 for authorizing SIP requests. I don't think the values of the attributes (=
as opposed to the mere presence of some attributes) have anything to do wit=
h authorizing the SIP request that you propose would carry the attributes.<=
/div>
<div><br>
</div>
<div>
<div>With regards to this whole conferencing example, it does seem a bit co=
ntrived to me, like you're sketching a SIP-based way of setting conference =
policy on the fly that happens to require the kind of attributes you want, =
and then saying that demonstrates
 the need for the attributes. The fact that all of the ways to do conferenc=
e policy we can point to in fact do it via the web makes it hard for me to =
understand why we should create a SIP-based way of doing this. To me, it lo=
oks like you're opening a problem
 space that has much more to do with reshaping the way that SIP does confer=
ences (and again, as Radhika's mail pointing out, there's been a lot of thi=
nking about this since the days of RFC4579) than it does with defining a fr=
amework for authorizing SIP requests.</div>
</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>Thursday, June 16, 2016 at 6:=
19 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>&quot;<a href=3D"mailto:sipcore=
@ietf.org">sipcore@ietf.org</a>&quot; &lt;<a href=3D"mailto:sipcore@ietf.or=
g">sipcore@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [sipcore] SIP AuthNZ P=
roblem Statement - v3<br>
</div>
<div><br>
</div>
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div>
<div>
<div dir=3D"ltr">
<div>The attributes I provided are examples of attributes that might be pro=
vided in an access token; it was not meant to include all the attributes fo=
r this use case, because my intention was not to focus on conference server=
, but to discuss the idea in general.</div>
<div><br>
</div>
<div>
<div>The following could be used with the conference factory to request a c=
onference focus:</div>
<div>The INVITE sent to the conference factory could include a token that h=
as a <b>
conference-type</b> attribute which could take one of the following values:=
 <b>audio, video, webconference, audio-webconference, or video-webconferenc=
e</b>.</div>
<div>The conference factory will then issue a <b>conference uri</b> for the=
 requested conference type, assuming the token is valid. If the
<b>conference-type</b> attribute is missing, then the request would be reje=
cted.</div>
</div>
<div><br>
</div>
<div>Is that what you are looking for?</div>
<div><br>
</div>
<div>Regards,</div>
<div>&nbsp;Rifaat</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Wed, Jun 15, 2016 at 5:06 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:0 0 0 .8ex;border-left:1p=
x #ccc 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 class=3D""><span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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">
<span><br>
<br>
</span>
<div>
<div>In that case, if you could furnish a use case with attributes that wou=
ld be carried in a SIP iNVITE that are actually salient to the authorizatio=
n of that particular SIP request, we'd be getting a lot closer to common gr=
ound.</div>
<div><br>
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>An obvious example would be an INVITE with a token sent to a conferenc=
e factory to authorize the creation of a focus.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span>
<div>We're still talking past each other. My point above was that the two a=
ttributes you specified - constraints on conference size and media types - =
have nothing to do with the authorization of the INVITE carrying those attr=
ibutes. Like, unless the conference
 size permitted is 0, then the INVITE that creates an ad hoc conference and=
 connects you to its bridge is not going to be rejected by the focus on the=
 basis of that conference size attribute. If the maximum conference size we=
re 0, the AuthZ service just wouldn't
 attach a token to an INVITE as it would never be allowed to start a confer=
ence. Same goes for media types.</div>
<div><br>
</div>
<div>Instead, what your two attributes constrain is what happens after the =
conference is created: when ten users try to dial in to a conference with a=
 permitted size of nine, say. That tenth INVITE would get rejected - not th=
e conference-forming INVITE that
 you propose would carry the token. That's why what you're proposing is ove=
rloading the INVITE to tunnel conference policy constraints to the focus. I=
 mean, maybe that's a problem space worth investigating, though I suspect t=
here are good reasons why RFC4353
 specified that conference policy is not manipulated with SIP - mostly beca=
use the semantics of manipulating conference policy probably don't map well=
 onto dialog-forming SIP requests. But I think that conference policy provi=
sioning problem space would be pretty
 far from where we started in this discussion.</div>
<div><br>
</div>
<div>Let me reiterate what I wrote in my previous mail, which I quote here =
at the top: &quot;furnish a use case with attributes that would be carried =
in a SIP iNVITE that are actually salient to the authorization of that part=
icular SIP request.&quot; Again: &quot;that particular
 SIP request.&quot; Not later transactions in some broader context in which=
 the original user agent probably isn't even an endpoint.</div>
<div><br>
</div>
<div>The Authorization header in SIP, for example, is actually used by the =
UAS to make a decision about whether or not to authorize the request in whi=
ch the header appears.&nbsp; The token that you propose a SIP INVITE would =
carry in this case is not used by the
 relying party to make a decision about whether or not that INVITE is autho=
rized. That's why I believe this conference policy issue is a different pro=
blem space entirely from figuring out how to implement attribute-based auth=
orization for SIP requests.</div>
<span class=3D"">
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<div>Regards,</div>
<div>&nbsp;Rifaat</div>
<div><br>
</div>
</div>
</div>
</div>
</blockquote>
</span></span></div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</blockquote>
</span>
</body>
</html>

--_000_D38F19C319F959jonpetersonneustarbiz_--


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From: Roman Shpount <roman@telurix.com>
Date: Tue, 21 Jun 2016 19:50:13 -0400
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>, Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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--94eb2c05abe8711c520535d27ba0
Content-Type: text/plain; charset=UTF-8

Furthermore, reading all of this, I cannot understand why Authorization
token cannot be a part of SIP URL, either as user part or URL parameter.
This does not need to be anything more then service specific URL and does
not require a new SIP header or authentication mechanism.

_____________
Roman Shpount

On Tue, Jun 21, 2016 at 7:46 PM, Peterson, Jon <jon.peterson@neustar.biz>
wrote:

>
> It's not really what I'm looking for, no, and after mulling this over a
> bit, I fear going farther along this path would only drill down into levels
> of complexity that will likely make this harder to understand rather than
> easier.
>
> I do acknowledge that you describe the INVITE as being rejected if it
> doesn't contain a token - but that's not the same thing as saying the
> INVITE will be rejected because the relying party makes an authorization
> decision based on the attributes in the token. It's more like an INVITE
> being rejected because a user agent can't supply an Authorization header. I
> will reiterate that I don't think the kinds of attributes you are proposing
> should be in the scope of an effort that will try to address a framework
> for authorizing SIP requests. I don't think the values of the attributes
> (as opposed to the mere presence of some attributes) have anything to do
> with authorizing the SIP request that you propose would carry the
> attributes.
>
> With regards to this whole conferencing example, it does seem a bit
> contrived to me, like you're sketching a SIP-based way of setting
> conference policy on the fly that happens to require the kind of attributes
> you want, and then saying that demonstrates the need for the attributes.
> The fact that all of the ways to do conference policy we can point to in
> fact do it via the web makes it hard for me to understand why we should
> create a SIP-based way of doing this. To me, it looks like you're opening a
> problem space that has much more to do with reshaping the way that SIP does
> conferences (and again, as Radhika's mail pointing out, there's been a lot
> of thinking about this since the days of RFC4579) than it does with
> defining a framework for authorizing SIP requests.
>
> Jon Peterson
> Neustar, Inc.
>
> From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
> Date: Thursday, June 16, 2016 at 6:19 AM
> To: Jon Peterson <jon.peterson@neustar.biz>
> Cc: "sipcore@ietf.org" <sipcore@ietf.org>
> Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
>
> The attributes I provided are examples of attributes that might be
> provided in an access token; it was not meant to include all the attributes
> for this use case, because my intention was not to focus on conference
> server, but to discuss the idea in general.
>
> The following could be used with the conference factory to request a
> conference focus:
> The INVITE sent to the conference factory could include a token that has a *
> conference-type* attribute which could take one of the following values: *audio,
> video, webconference, audio-webconference, or video-webconference*.
> The conference factory will then issue a *conference uri* for the
> requested conference type, assuming the token is valid. If the
> *conference-type* attribute is missing, then the request would be
> rejected.
>
> Is that what you are looking for?
>
> Regards,
>  Rifaat
>
>
>
>
>
> On Wed, Jun 15, 2016 at 5:06 PM, Peterson, Jon <jon.peterson@neustar.biz>
> wrote:
>
>>
>>>
>>> In that case, if you could furnish a use case with attributes that would
>>> be carried in a SIP iNVITE that are actually salient to the authorization
>>> of that particular SIP request, we'd be getting a lot closer to common
>>> ground.
>>>
>>>
>> An obvious example would be an INVITE with a token sent to a conference
>> factory to authorize the creation of a focus.
>>
>>
>> We're still talking past each other. My point above was that the two
>> attributes you specified - constraints on conference size and media types -
>> have nothing to do with the authorization of the INVITE carrying those
>> attributes. Like, unless the conference size permitted is 0, then the
>> INVITE that creates an ad hoc conference and connects you to its bridge is
>> not going to be rejected by the focus on the basis of that conference size
>> attribute. If the maximum conference size were 0, the AuthZ service just
>> wouldn't attach a token to an INVITE as it would never be allowed to start
>> a conference. Same goes for media types.
>>
>> Instead, what your two attributes constrain is what happens after the
>> conference is created: when ten users try to dial in to a conference with a
>> permitted size of nine, say. That tenth INVITE would get rejected - not the
>> conference-forming INVITE that you propose would carry the token. That's
>> why what you're proposing is overloading the INVITE to tunnel conference
>> policy constraints to the focus. I mean, maybe that's a problem space worth
>> investigating, though I suspect there are good reasons why RFC4353
>> specified that conference policy is not manipulated with SIP - mostly
>> because the semantics of manipulating conference policy probably don't map
>> well onto dialog-forming SIP requests. But I think that conference policy
>> provisioning problem space would be pretty far from where we started in
>> this discussion.
>>
>> Let me reiterate what I wrote in my previous mail, which I quote here at
>> the top: "furnish a use case with attributes that would be carried in a SIP
>> iNVITE that are actually salient to the authorization of that particular
>> SIP request." Again: "that particular SIP request." Not later transactions
>> in some broader context in which the original user agent probably isn't
>> even an endpoint.
>>
>> The Authorization header in SIP, for example, is actually used by the UAS
>> to make a decision about whether or not to authorize the request in which
>> the header appears.  The token that you propose a SIP INVITE would carry in
>> this case is not used by the relying party to make a decision about whether
>> or not that INVITE is authorized. That's why I believe this conference
>> policy issue is a different problem space entirely from figuring out how to
>> implement attribute-based authorization for SIP requests.
>>
>> Jon Peterson
>> Neustar, Inc.
>>
>>
>> Regards,
>>  Rifaat
>>
>>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>
>

--94eb2c05abe8711c520535d27ba0
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Furthermore, reading all of this, I cannot understand why =
Authorization token cannot be a part of SIP URL, either as user part or URL=
 parameter. This does not need to be anything more then service specific UR=
L and does not require a new SIP header or authentication mechanism.</div><=
div class=3D"gmail_extra"><br clear=3D"all"><div><div class=3D"gmail_signat=
ure" data-smartmail=3D"gmail_signature">_____________<br>Roman Shpount</div=
></div>
<br><div class=3D"gmail_quote">On Tue, Jun 21, 2016 at 7:46 PM, Peterson, J=
on <span dir=3D"ltr">&lt;<a href=3D"mailto:jon.peterson@neustar.biz" target=
=3D"_blank">jon.peterson@neustar.biz</a>&gt;</span> wrote:<br><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc 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>It&#39;s not really what I&#39;m looking for, no, and after mulling th=
is over a bit, I fear going farther along this path would only drill down i=
nto levels of complexity that will likely make this harder to understand ra=
ther than easier.</div>
<div><br>
</div>
<div>I do acknowledge that you describe the INVITE as being rejected if it =
doesn&#39;t contain a token - but that&#39;s not the same thing as saying t=
he INVITE will be rejected because the relying party makes an authorization=
 decision based on the attributes in the
 token. It&#39;s more like an INVITE being rejected because a user agent ca=
n&#39;t supply an Authorization header. I will reiterate that I don&#39;t t=
hink the kinds of attributes you are proposing should be in the scope of an=
 effort that will try to address a framework
 for authorizing SIP requests. I don&#39;t think the values of the attribut=
es (as opposed to the mere presence of some attributes) have anything to do=
 with authorizing the SIP request that you propose would carry the attribut=
es.</div>
<div><br>
</div>
<div>
<div>With regards to this whole conferencing example, it does seem a bit co=
ntrived to me, like you&#39;re sketching a SIP-based way of setting confere=
nce policy on the fly that happens to require the kind of attributes you wa=
nt, and then saying that demonstrates
 the need for the attributes. The fact that all of the ways to do conferenc=
e policy we can point to in fact do it via the web makes it hard for me to =
understand why we should create a SIP-based way of doing this. To me, it lo=
oks like you&#39;re opening a problem
 space that has much more to do with reshaping the way that SIP does confer=
ences (and again, as Radhika&#39;s mail pointing out, there&#39;s been a lo=
t of thinking about this since the days of RFC4579) than it does with defin=
ing a framework for authorizing SIP requests.</div>
</div>
<div><br>
</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-BOTTOM:medium none;BORDER-LEFT:medium none;PADDING-BOTTOM:0in;PADD=
ING-LEFT:0in;PADDING-RIGHT:0in;BORDER-TOP:#b5c4df 1pt solid;BORDER-RIGHT:me=
dium 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" target=3D"_blank">rifaat.ietf@gmail.com<=
/a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Thursday, June 16, 2016 at 6:=
19 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>&quot;<a href=3D"mailto:sipcore=
@ietf.org" target=3D"_blank">sipcore@ietf.org</a>&quot; &lt;<a href=3D"mail=
to:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [sipcore] SIP AuthNZ P=
roblem Statement - v3<br>
</div>
<div><br>
</div>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div>
<div>
<div dir=3D"ltr">
<div>The attributes I provided are examples of attributes that might be pro=
vided in an access token; it was not meant to include all the attributes fo=
r this use case, because my intention was not to focus on conference server=
, but to discuss the idea in general.</div>
<div><br>
</div>
<div>
<div>The following could be used with the conference factory to request a c=
onference focus:</div>
<div>The INVITE sent to the conference factory could include a token that h=
as a <b>
conference-type</b> attribute which could take one of the following values:=
 <b>audio, video, webconference, audio-webconference, or video-webconferenc=
e</b>.</div>
<div>The conference factory will then issue a <b>conference uri</b> for the=
 requested conference type, assuming the token is valid. If the
<b>conference-type</b> attribute is missing, then the request would be reje=
cted.</div>
</div>
<div><br>
</div>
<div>Is that what you are looking for?</div>
<div><br>
</div>
<div>Regards,</div>
<div>=C2=A0Rifaat</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Wed, Jun 15, 2016 at 5:06 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:0 0 0 .8ex;border-left:1p=
x #ccc 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><span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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">
<span><br>
<br>
</span>
<div>
<div>In that case, if you could furnish a use case with attributes that wou=
ld be carried in a SIP iNVITE that are actually salient to the authorizatio=
n of that particular SIP request, we&#39;d be getting a lot closer to commo=
n ground.</div>
<div><br>
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>An obvious example would be an INVITE with a token sent to a conferenc=
e factory to authorize the creation of a focus.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span>
<div>We&#39;re still talking past each other. My point above was that the t=
wo attributes you specified - constraints on conference size and media type=
s - have nothing to do with the authorization of the INVITE carrying those =
attributes. Like, unless the conference
 size permitted is 0, then the INVITE that creates an ad hoc conference and=
 connects you to its bridge is not going to be rejected by the focus on the=
 basis of that conference size attribute. If the maximum conference size we=
re 0, the AuthZ service just wouldn&#39;t
 attach a token to an INVITE as it would never be allowed to start a confer=
ence. Same goes for media types.</div>
<div><br>
</div>
<div>Instead, what your two attributes constrain is what happens after the =
conference is created: when ten users try to dial in to a conference with a=
 permitted size of nine, say. That tenth INVITE would get rejected - not th=
e conference-forming INVITE that
 you propose would carry the token. That&#39;s why what you&#39;re proposin=
g is overloading the INVITE to tunnel conference policy constraints to the =
focus. I mean, maybe that&#39;s a problem space worth investigating, though=
 I suspect there are good reasons why RFC4353
 specified that conference policy is not manipulated with SIP - mostly beca=
use the semantics of manipulating conference policy probably don&#39;t map =
well onto dialog-forming SIP requests. But I think that conference policy p=
rovisioning problem space would be pretty
 far from where we started in this discussion.</div>
<div><br>
</div>
<div>Let me reiterate what I wrote in my previous mail, which I quote here =
at the top: &quot;furnish a use case with attributes that would be carried =
in a SIP iNVITE that are actually salient to the authorization of that part=
icular SIP request.&quot; Again: &quot;that particular
 SIP request.&quot; Not later transactions in some broader context in which=
 the original user agent probably isn&#39;t even an endpoint.</div>
<div><br>
</div>
<div>The Authorization header in SIP, for example, is actually used by the =
UAS to make a decision about whether or not to authorize the request in whi=
ch the header appears.=C2=A0 The token that you propose a SIP INVITE would =
carry in this case is not used by the
 relying party to make a decision about whether or not that INVITE is autho=
rized. That&#39;s why I believe this conference policy issue is a different=
 problem space entirely from figuring out how to implement attribute-based =
authorization for SIP requests.</div>
<span>
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<div>Regards,</div>
<div>=C2=A0Rifaat</div>
<div><br>
</div>
</div>
</div>
</div>
</blockquote>
</span></span></div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</blockquote>
</span>
</div>

<br>_______________________________________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" rel=3D"noreferrer=
" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sipcore</a><br>
<br></blockquote></div><br></div>

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Gonzalo Salgueiro (gsalguei)" <gsalguei@cisco.com>, Ben Campbell <ben@nostrum.com>
Thread-Topic: AD Evaluation of draft-holmberg-dispatch-rfc7315-updates-06
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Subject: Re: [sipcore] AD Evaluation of draft-holmberg-dispatch-rfc7315-updates-06
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Hi,

We can add the text.

Regards,

Christer

Sent from my Windows Phone
________________________________
From: Gonzalo Salgueiro (gsalguei)<mailto:gsalguei@cisco.com>
Sent: =FD21/=FD06/=FD2016 19:44
To: Ben Campbell<mailto:ben@nostrum.com>
Cc: Christer Holmberg<mailto:christer.holmberg@ericsson.com>; draft-holmber=
g-dispatch-rfc7315-updates.all@ietf.org<mailto:draft-holmberg-dispatch-rfc7=
315-updates.all@ietf.org>; SIPCORE<mailto:sipcore@ietf.org>
Subject: Re: AD Evaluation of draft-holmberg-dispatch-rfc7315-updates-06


> On Jun 21, 2016, at 11:22 AM, Ben Campbell <ben@nostrum.com> wrote:
>
> That's a good start, but don't be surprised if we get questions specifica=
lly about adding NPLI to ACK requests. some language to the effect of the f=
ollowing might help:
>
> "This document adds the ability to include P-Access-Network-Info in ACK r=
equests. As documented in RFC7315, P-Access-Network-Info may include privac=
y sensitive information, including the user's location. The security and pr=
ivacy considerations for P-Access-Network-Info in ACK requests are similar =
to those for the other SIP requests discussed in RFC7315.=94

I=92m fine with adding such text.

Christer - Can we append this to your proposed text?

Gonzalo


>
> Thanks!
>
> Ben.
>
> On 21 Jun 2016, at 3:26, Christer Holmberg wrote:
>
>> Hi Ben,
>>
>> See inline.
>>
>>> --------------
>>>
>>> Substantive:
>>>
>>> The security considerations state that the draft removes some places
>>> that some of the P-Headers can be sent, but expands that to some other
>>> places. Further, it says that neither introduce new security
>>> considerations beyond those in 7315.
>>>
>>> I accept that for the reduction part. But I'm not sure we can state tha=
t
>>> sort of thing for the expansion part, at least without some more
>>> discussion. Since 7315 already acknowledges potential privacy issues
>>> around P-Access-Network-Info, I'd like to at least see a sentence or tw=
o
>>> about the allowance of that in ACK requests, even if they just say that
>>> this addition makes things no worse than they already are.
>>
>>
>> OLD:
>>
>> The security considerations for P- header fields are defined in
>>   [RFC7315].  This specification allows some header fields to be
>>   present in messages where they were previously not allowed, and
>>   disallow some header fields to be present in messages where they were
>>   previously allowed. That does not cause any security issues, but
>>   implementations need to be aware that implementations may not have
>>   been updated according to this document, and take proper actions if a
>>   header field occur, or does not occur, in a message where it should
>>   occur (or occurs in a message where it should not occur).
>>
>>
>>
>> NEW:
>>
>> The security considerations for these P- header fields are defined in
>>   [RFC7315].  This specification allows some header fields to be
>>   present in messages where they were previously not allowed, and the
>> security considerations and assumptions (e.g. regarding only sending
>> Information to trusted entities) also to those messages. In addition,
>> this specification also disallow some header fields to be present
>> in message where they were previously allowed. That does not cause
>> any security issues, but implementations need to be aware that
>> implementations may not have been updated according to this document,
>> and take proper actions if a header field occur, or does not occur,
>> in a message where it should occur (or occurs in a message where it
>> should not occur).
>>
>>
>>
>>> Editorial:
>>>
>>> -5, first sentence: "The security considerations for P- header fields
>>> are defined in
>>>   [RFC7315]"
>>> I assume this means 7315 discusses the security considerations for thes=
e
>>> P-Headers specifically, not P-Headers in general. Is this the intent? I=
f
>>> so, I suggest:
>>>
>>> s/... for P-header fields.../ ... for these P-header fields...
>>
>> I=B9ll fix as suggested (ass new text above).
>>
>> Regards,
>>
>> Christer


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</head>
<body>
<div>
<div>
<div style=3D"font-family:Calibri,sans-serif; font-size:11pt">Hi,<br>
<br>
We can add the text.<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:gsalguei@cisco.com">Gonzalo Salgueiro (gsalguei)</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">=FD21=
/=FD06/=FD2016 19:44</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:ben@nostrum.com">Ben Campbell</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Cc:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:christer.holmberg@ericsson.com">Christer Holmberg</a>;
<a href=3D"mailto:draft-holmberg-dispatch-rfc7315-updates.all@ietf.org">dra=
ft-holmberg-dispatch-rfc7315-updates.all@ietf.org</a>;
<a href=3D"mailto:sipcore@ietf.org">SIPCORE</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: A=
D Evaluation of draft-holmberg-dispatch-rfc7315-updates-06</span><br>
<br>
</div>
</div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText"><br>
&gt; On Jun 21, 2016, at 11:22 AM, Ben Campbell &lt;ben@nostrum.com&gt; wro=
te:<br>
&gt; <br>
&gt; That's a good start, but don't be surprised if we get questions specif=
ically about adding NPLI to ACK requests. some language to the effect of th=
e following might help:<br>
&gt; <br>
&gt; &quot;This document adds the ability to include P-Access-Network-Info =
in ACK requests. As documented in RFC7315, P-Access-Network-Info may includ=
e privacy sensitive information, including the user's location. The securit=
y and privacy considerations for P-Access-Network-Info
 in ACK requests are similar to those for the other SIP requests discussed =
in RFC7315.=94<br>
<br>
I=92m fine with adding such text.<br>
<br>
Christer - Can we append this to your proposed text?<br>
<br>
Gonzalo<br>
<br>
<br>
&gt; <br>
&gt; Thanks!<br>
&gt; <br>
&gt; Ben.<br>
&gt; <br>
&gt; On 21 Jun 2016, at 3:26, Christer Holmberg wrote:<br>
&gt; <br>
&gt;&gt; Hi Ben,<br>
&gt;&gt; <br>
&gt;&gt; See inline.<br>
&gt;&gt; <br>
&gt;&gt;&gt; --------------<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; Substantive:<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; The security considerations state that the draft removes some =
places<br>
&gt;&gt;&gt; that some of the P-Headers can be sent, but expands that to so=
me other<br>
&gt;&gt;&gt; places. Further, it says that neither introduce new security<b=
r>
&gt;&gt;&gt; considerations beyond those in 7315.<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; I accept that for the reduction part. But I'm not sure we can =
state that<br>
&gt;&gt;&gt; sort of thing for the expansion part, at least without some mo=
re<br>
&gt;&gt;&gt; discussion. Since 7315 already acknowledges potential privacy =
issues<br>
&gt;&gt;&gt; around P-Access-Network-Info, I'd like to at least see a sente=
nce or two<br>
&gt;&gt;&gt; about the allowance of that in ACK requests, even if they just=
 say that<br>
&gt;&gt;&gt; this addition makes things no worse than they already are.<br>
&gt;&gt; <br>
&gt;&gt; <br>
&gt;&gt; OLD:<br>
&gt;&gt; <br>
&gt;&gt; The security considerations for P- header fields are defined in<br=
>
&gt;&gt;&nbsp;&nbsp; [RFC7315].&nbsp; This specification allows some header=
 fields to be<br>
&gt;&gt;&nbsp;&nbsp; present in messages where they were previously not all=
owed, and<br>
&gt;&gt;&nbsp;&nbsp; disallow some header fields to be present in messages =
where they were<br>
&gt;&gt;&nbsp;&nbsp; previously allowed. That does not cause any security i=
ssues, but<br>
&gt;&gt;&nbsp;&nbsp; implementations need to be aware that implementations =
may not have<br>
&gt;&gt;&nbsp;&nbsp; been updated according to this document, and take prop=
er actions if a<br>
&gt;&gt;&nbsp;&nbsp; header field occur, or does not occur, in a message wh=
ere it should<br>
&gt;&gt;&nbsp;&nbsp; occur (or occurs in a message where it should not occu=
r).<br>
&gt;&gt; <br>
&gt;&gt; <br>
&gt;&gt; <br>
&gt;&gt; NEW:<br>
&gt;&gt; <br>
&gt;&gt; The security considerations for these P- header fields are defined=
 in<br>
&gt;&gt;&nbsp;&nbsp; [RFC7315].&nbsp; This specification allows some header=
 fields to be<br>
&gt;&gt;&nbsp;&nbsp; present in messages where they were previously not all=
owed, and the<br>
&gt;&gt; security considerations and assumptions (e.g. regarding only sendi=
ng<br>
&gt;&gt; Information to trusted entities) also to those messages. In additi=
on,<br>
&gt;&gt; this specification also disallow some header fields to be present<=
br>
&gt;&gt; in message where they were previously allowed. That does not cause=
<br>
&gt;&gt; any security issues, but implementations need to be aware that<br>
&gt;&gt; implementations may not have been updated according to this docume=
nt,<br>
&gt;&gt; and take proper actions if a header field occur, or does not occur=
,<br>
&gt;&gt; in a message where it should occur (or occurs in a message where i=
t<br>
&gt;&gt; should not occur).<br>
&gt;&gt; <br>
&gt;&gt; <br>
&gt;&gt; <br>
&gt;&gt;&gt; Editorial:<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; -5, first sentence: &quot;The security considerations for P- h=
eader fields<br>
&gt;&gt;&gt; are defined in<br>
&gt;&gt;&gt;&nbsp;&nbsp; [RFC7315]&quot;<br>
&gt;&gt;&gt; I assume this means 7315 discusses the security considerations=
 for these<br>
&gt;&gt;&gt; P-Headers specifically, not P-Headers in general. Is this the =
intent? If<br>
&gt;&gt;&gt; so, I suggest:<br>
&gt;&gt;&gt; <br>
&gt;&gt;&gt; s/... for P-header fields.../ ... for these P-header fields...=
<br>
&gt;&gt; <br>
&gt;&gt; I=B9ll fix as suggested (ass new text above).<br>
&gt;&gt; <br>
&gt;&gt; Regards,<br>
&gt;&gt; <br>
&gt;&gt; Christer<br>
<br>
</div>
</span></font>
</body>
</html>

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "Gonzalo Salgueiro (gsalguei)" <gsalguei@cisco.com>, Ben Campbell <ben@nostrum.com>
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References: <CAGL6ep+8nhiuQs8uBLxmsdwfrxFys4PM8UP5Apbcb7YVK4fmfg@mail.gmail.com> <CAGL6epKhbrdNEgpLFDD+JHFCNcAfST1aCKg3votcVe6z4wVa6A@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B37F97548@ESESSMB209.ericsson.se> <D3562523.18AAA1%jon.peterson@neustar.biz> <CAGL6epJDu1ojsSaPnJqpNoZgCwXvOoFNoEa2ZLNtmRtKMPMyoA@mail.gmail.com> <D35B4CE3.18D9AB%jon.peterson@neustar.biz> <CAGL6epJuByqHi_Fvssyd0V-LE4=_s7HK9DB-9Z3SSVky1Ak4Xw@mail.gmail.com> <D361F10A.18F366%jon.peterson@neustar.biz> <CAGL6epKVaOyVMS_6xmdEXbWsp2UXsbbwjTW2MgM=fJWLE381dg@mail.gmail.com> <D3688271.191B87%jon.peterson@neustar.biz> <CAGL6epKiY_Xi_D+J8QnTfoEyaDU1=jfcbhVFAMY_v3tkQjAhug@mail.gmail.com> <D3842CA2.19BA97%jon.peterson@neustar.biz> <CAGL6epKbdjHP-K9Hjn825HUVTQ4V=TnUYeyrLSU0ETQtmsfs9g@mail.gmail.com> <D385862C.19C10C%jon.peterson@neustar.biz> <CAGL6epLZOhUMQHGGW_Y49gZZAdvFWH07NfO97f=krk1_K9Xd-g@mail.gmail.com> <D386CD3B.19C95F%jon.peterson@neustar.biz> <CAGL6epKj00is8dH7NHdjjuTbh-XbeB7yuALouzR2fzfMBC=R4w@mail.gmail.com> <D38F19C3.19F959%jon.peterson@neustar.biz>
From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Date: Wed, 22 Jun 2016 08:41:19 -0400
Message-ID: <CAGL6epK_DEPr5nYOfGOszLtfT8i8Lso8cZkdXO2+v613ix3eDQ@mail.gmail.com>
To: "Peterson, Jon" <jon.peterson@neustar.biz>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/sipcore/ypo2xHv6EVVbb6ZVNLaF8NIMx0c>
Cc: "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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Jon,



I think we are still talking past each other.



The token added to the INVITE is *not a policy*; it is an authorization by
the AuthZ Server to allow this request to create a conference focus.

The policy is created and managed in the AuthZ Server; remember that the
idea is to delegate the AuthZ to a separate entity other than the server
providing the service.





Let=E2=80=99s continue with the conference use case:


     Trust Domain 1              :    Internet     :  Trust Domain 2

                                 :                 :

                  +--------+     :                 :

                  | IdP    |     :                 :

    +-------------| AuthZ  |     :                 :

    |             |        |     :                 :

    |             +--------+     :                 :

    |                 |          :                 :

    |                 |          :                 :

+--------+        +--------+     :                 :     +--------+

|        |        | SIP    |     :                 :     | SIP App|

|   UA   |--------| Proxy  |-----:-----------------:-----| Video  |

|        |        |        |     :                 :     | Conference

+--------+        +--------+     :                 :     +--------+

                                 :                 :



Let=E2=80=99s assume that the domain in Trust Domain 1 is example.com, and =
Alice
needs an account with this domain and needs access to the conference server=
.



The administrator of the example.com domain creates an account for Alice (
alice@example.com) and sets her *conference policy* to =E2=80=9C*conference=
-type **=3D
audio | audio-webconference=E2=80=9D* to allow her to create either an audi=
o
conference or an audio-webconference.



Now, Alice decides to start an audio conference; to do that, the phone
obtains an access token from the AuthNZ server that allows Alice to start a
conference of the type =E2=80=9C*audio=E2=80=9D*.



When Alice sends the INVITE to the conference server, the phone will
include that specific access token, which will be used by the conference
factory to authorize the request to create a conference focus.



So, the conference server has no idea what the policy is; all it knows is
that it received a request with a token created by a specific AuthNZ
server. If the conference server trusts the AuthNZ server and the token is
valid, it authorizes the request and creates the requested conference
focus; otherwise, the request is rejected.



Hopefully this is clearer.



Regards,

 Rifaat



















On Tue, Jun 21, 2016 at 7:46 PM, Peterson, Jon <jon.peterson@neustar.biz>
wrote:

>
> It's not really what I'm looking for, no, and after mulling this over a
> bit, I fear going farther along this path would only drill down into leve=
ls
> of complexity that will likely make this harder to understand rather than
> easier.
>
> I do acknowledge that you describe the INVITE as being rejected if it
> doesn't contain a token - but that's not the same thing as saying the
> INVITE will be rejected because the relying party makes an authorization
> decision based on the attributes in the token. It's more like an INVITE
> being rejected because a user agent can't supply an Authorization header.=
 I
> will reiterate that I don't think the kinds of attributes you are proposi=
ng
> should be in the scope of an effort that will try to address a framework
> for authorizing SIP requests. I don't think the values of the attributes
> (as opposed to the mere presence of some attributes) have anything to do
> with authorizing the SIP request that you propose would carry the
> attributes.
>
> With regards to this whole conferencing example, it does seem a bit
> contrived to me, like you're sketching a SIP-based way of setting
> conference policy on the fly that happens to require the kind of attribut=
es
> you want, and then saying that demonstrates the need for the attributes.
> The fact that all of the ways to do conference policy we can point to in
> fact do it via the web makes it hard for me to understand why we should
> create a SIP-based way of doing this. To me, it looks like you're opening=
 a
> problem space that has much more to do with reshaping the way that SIP do=
es
> conferences (and again, as Radhika's mail pointing out, there's been a lo=
t
> of thinking about this since the days of RFC4579) than it does with
> defining a framework for authorizing SIP requests.
>
> Jon Peterson
> Neustar, Inc.
>
> From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
> Date: Thursday, June 16, 2016 at 6:19 AM
> To: Jon Peterson <jon.peterson@neustar.biz>
> Cc: "sipcore@ietf.org" <sipcore@ietf.org>
> Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
>
> The attributes I provided are examples of attributes that might be
> provided in an access token; it was not meant to include all the attribut=
es
> for this use case, because my intention was not to focus on conference
> server, but to discuss the idea in general.
>
> The following could be used with the conference factory to request a
> conference focus:
> The INVITE sent to the conference factory could include a token that has =
a *
> conference-type* attribute which could take one of the following values: =
*audio,
> video, webconference, audio-webconference, or video-webconference*.
> The conference factory will then issue a *conference uri* for the
> requested conference type, assuming the token is valid. If the
> *conference-type* attribute is missing, then the request would be
> rejected.
>
> Is that what you are looking for?
>
> Regards,
>  Rifaat
>
>
>
>
>
> On Wed, Jun 15, 2016 at 5:06 PM, Peterson, Jon <jon.peterson@neustar.biz>
> wrote:
>
>>
>>>
>>> In that case, if you could furnish a use case with attributes that woul=
d
>>> be carried in a SIP iNVITE that are actually salient to the authorizati=
on
>>> of that particular SIP request, we'd be getting a lot closer to common
>>> ground.
>>>
>>>
>> An obvious example would be an INVITE with a token sent to a conference
>> factory to authorize the creation of a focus.
>>
>>
>> We're still talking past each other. My point above was that the two
>> attributes you specified - constraints on conference size and media type=
s -
>> have nothing to do with the authorization of the INVITE carrying those
>> attributes. Like, unless the conference size permitted is 0, then the
>> INVITE that creates an ad hoc conference and connects you to its bridge =
is
>> not going to be rejected by the focus on the basis of that conference si=
ze
>> attribute. If the maximum conference size were 0, the AuthZ service just
>> wouldn't attach a token to an INVITE as it would never be allowed to sta=
rt
>> a conference. Same goes for media types.
>>
>> Instead, what your two attributes constrain is what happens after the
>> conference is created: when ten users try to dial in to a conference wit=
h a
>> permitted size of nine, say. That tenth INVITE would get rejected - not =
the
>> conference-forming INVITE that you propose would carry the token. That's
>> why what you're proposing is overloading the INVITE to tunnel conference
>> policy constraints to the focus. I mean, maybe that's a problem space wo=
rth
>> investigating, though I suspect there are good reasons why RFC4353
>> specified that conference policy is not manipulated with SIP - mostly
>> because the semantics of manipulating conference policy probably don't m=
ap
>> well onto dialog-forming SIP requests. But I think that conference polic=
y
>> provisioning problem space would be pretty far from where we started in
>> this discussion.
>>
>> Let me reiterate what I wrote in my previous mail, which I quote here at
>> the top: "furnish a use case with attributes that would be carried in a =
SIP
>> iNVITE that are actually salient to the authorization of that particular
>> SIP request." Again: "that particular SIP request." Not later transactio=
ns
>> in some broader context in which the original user agent probably isn't
>> even an endpoint.
>>
>> The Authorization header in SIP, for example, is actually used by the UA=
S
>> to make a decision about whether or not to authorize the request in whic=
h
>> the header appears.  The token that you propose a SIP INVITE would carry=
 in
>> this case is not used by the relying party to make a decision about whet=
her
>> or not that INVITE is authorized. That's why I believe this conference
>> policy issue is a different problem space entirely from figuring out how=
 to
>> implement attribute-based authorization for SIP requests.
>>
>> Jon Peterson
>> Neustar, Inc.
>>
>>
>> Regards,
>>  Rifaat
>>
>>
>

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<div dir=3D"ltr"><p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><s=
pan style=3D"font-size:10pt;line-height:115%"><font face=3D"monospace, mono=
space">Jon,</font></span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%"><font face=3D"monospace, monospace">=C2=A0</fo=
nt></span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%"><font face=3D"monospace, monospace">I think we=
 are
still talking past each other.</font></span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%"><font face=3D"monospace, monospace">=C2=A0</fo=
nt></span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%"><font face=3D"monospace, monospace">The token =
added
to the INVITE is <b>not a policy</b>; it is
an authorization by the AuthZ Server to allow this request to create a
conference focus.</font></span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%"><font face=3D"monospace, monospace">The policy=
 is
created and managed in the AuthZ Server; remember that the idea is to deleg=
ate
the AuthZ to a separate entity other than the server providing the service.=
</font></span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%"><font face=3D"monospace, monospace">=C2=A0</fo=
nt></span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%"><font face=3D"monospace, monospace">=C2=A0</fo=
nt></span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%"><font face=3D"monospace, monospace">Let=E2=80=
=99s continue
with the conference use case:</font></span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial"><font face=3D"monospace, monospace"><span s=
tyle=3D"font-size:10pt"><br>
=C2=A0 =C2=A0 =C2=A0Trust Domain 1 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0
=C2=A0: =C2=A0 =C2=A0Internet =C2=A0 =C2=A0 : =C2=A0Trust Domain 2</span><s=
pan style=3D"font-size:10pt"></span></font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial"><font face=3D"monospace, monospace"><span s=
tyle=3D"font-size:10pt">=C2=A0
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0
=C2=A0 =C2=A0 :</span><span style=3D"font-size:10pt"></span></font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial"><font face=3D"monospace, monospace"><span s=
tyle=3D"font-size:10pt">=C2=A0
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 +--------+ =C2=A0
=C2=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :</span><s=
pan style=3D"font-size:10pt"></span></font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial"><font face=3D"monospace, monospace"><span s=
tyle=3D"font-size:10pt">=C2=A0
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | IdP =C2=A0 =C2=A0=
|
=C2=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :</=
span><span style=3D"font-size:10pt"></span></font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial"><font face=3D"monospace, monospace"><span s=
tyle=3D"font-size:10pt">=C2=A0
=C2=A0 +-------------| AuthZ =C2=A0| =C2=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=A0
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :</span><span style=3D"font-size:10pt"><=
/span></font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial"><font face=3D"monospace, monospace"><span s=
tyle=3D"font-size:10pt">=C2=A0
=C2=A0 | =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0
=C2=A0| =C2=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0
:</span><span style=3D"font-size:10pt"></span></font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial"><font face=3D"monospace, monospace"><span s=
tyle=3D"font-size:10pt">=C2=A0
=C2=A0 | =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 +--------+ =C2=A0 =C2=A0=
 :
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :</span><span style=
=3D"font-size:10pt"></span></font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial"><font face=3D"monospace, monospace"><span s=
tyle=3D"font-size:10pt">=C2=A0
=C2=A0 | =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0
=C2=A0 =C2=A0 =C2=A0 =C2=A0: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0
=C2=A0 :</span><span style=3D"font-size:10pt"></span></font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial"><font face=3D"monospace, monospace"><span s=
tyle=3D"font-size:10pt">=C2=A0
=C2=A0 | =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0
=C2=A0 =C2=A0 =C2=A0 =C2=A0: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0
=C2=A0 :</span><span style=3D"font-size:10pt"></span></font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial"><font face=3D"monospace, monospace"><span s=
tyle=3D"font-size:10pt">+--------+
=C2=A0 =C2=A0 =C2=A0 =C2=A0+--------+ =C2=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=A0
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : =C2=A0 =C2=A0 +--------+</span><span s=
tyle=3D"font-size:10pt"></span></font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial"><font face=3D"monospace, monospace"><span s=
tyle=3D"font-size:10pt">|
=C2=A0 =C2=A0 =C2=A0 =C2=A0| =C2=A0 =C2=A0 =C2=A0 =C2=A0| SIP =C2=A0 =C2=A0=
|
=C2=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :
=C2=A0 =C2=A0 | SIP App|</span><span style=3D"font-size:10pt"></span></font=
></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial"><font face=3D"monospace, monospace"><span s=
tyle=3D"font-size:10pt">|
=C2=A0 UA =C2=A0 |--------| Proxy =C2=A0|-----:-----------------:-----| Vid=
eo
=C2=A0|</span><span style=3D"font-size:10pt"></span></font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial"><font face=3D"monospace, monospace"><span s=
tyle=3D"font-size:10pt">|
=C2=A0 =C2=A0 =C2=A0 =C2=A0| =C2=A0 =C2=A0 =C2=A0 =C2=A0| =C2=A0 =C2=A0 =C2=
=A0
=C2=A0| =C2=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0
: =C2=A0 =C2=A0 | Conference</span><span style=3D"font-size:10pt"></span></=
font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial"><font face=3D"monospace, monospace"><span s=
tyle=3D"font-size:10pt">+--------+
=C2=A0 =C2=A0 =C2=A0 =C2=A0+--------+ =C2=A0 =C2=A0 : =C2=A0 =C2=A0 =C2=A0
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : =C2=A0 =C2=A0 +--------+</span><span s=
tyle=3D"font-size:10pt"></span></font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt;background-image:ini=
tial;background-repeat:initial"><font face=3D"monospace, monospace"><span s=
tyle=3D"font-size:10pt">=C2=A0
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0
=C2=A0 =C2=A0 :</span><span style=3D"font-size:10pt"></span></font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%"><font face=3D"monospace, monospace">=C2=A0</fo=
nt></span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%"><font face=3D"monospace, monospace">Let=E2=80=
=99s assume
that the domain in Trust Domain 1 is <a href=3D"http://example.com">example=
.com</a>, and Alice needs an account with
this domain and needs access to the conference server.</font></span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%"><font face=3D"monospace, monospace">=C2=A0</fo=
nt></span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><font face=3D"monos=
pace, monospace"><span style=3D"font-size:10pt;line-height:115%">The
administrator of the <a href=3D"http://example.com">example.com</a> domain =
creates an account for Alice (<a href=3D"mailto:alice@example.com"><span st=
yle=3D"color:windowtext">alice@example.com</span></a>)
and sets her <b>conference policy</b> to =E2=80=9C</span><b><span style=3D"=
font-size:10pt;line-height:115%;background-image:initial;background-repeat:=
initial">conference-type </span></b><span style=3D"font-size:10pt;line-heig=
ht:115%;background-image:initial;background-repeat:initial"><b>=3D audio | =
audio-webconference=E2=80=9D</b> to allow
her to create either an audio conference or an audio-webconference.</span><=
/font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%;background-image:initial;background-repeat:init=
ial"><font face=3D"monospace, monospace">=C2=A0</font></span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%;background-image:initial;background-repeat:init=
ial"><font face=3D"monospace, monospace">Now, Alice decides to start
an audio conference; to do that, the phone obtains an access token from the
AuthNZ server that allows Alice to start a conference of the type =E2=80=9C=
<b>audio=E2=80=9D</b>.</font></span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%;background-image:initial;background-repeat:init=
ial"><font face=3D"monospace, monospace">=C2=A0</font></span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%;background-image:initial;background-repeat:init=
ial"><font face=3D"monospace, monospace">When Alice sends the INVITE
to the conference server, the phone will include that specific access token=
,
which will be used by the conference factory to authorize the request to cr=
eate
a conference focus. </font></span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%;background-image:initial;background-repeat:init=
ial"><font face=3D"monospace, monospace">=C2=A0</font></span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%;background-image:initial;background-repeat:init=
ial"><font face=3D"monospace, monospace">So, the conference server has
no idea what the policy is; all it knows is that it received a request with=
 a token
created by a specific AuthNZ server. If the conference server trusts the Au=
thNZ
server and the token is valid, it authorizes the request and creates the
requested conference focus; otherwise, the request is rejected.</font></spa=
n></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%;background-image:initial;background-repeat:init=
ial"><font face=3D"monospace, monospace">=C2=A0</font></span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%;background-image:initial;background-repeat:init=
ial"><font face=3D"monospace, monospace">Hopefully this is clearer.</font><=
/span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%;background-image:initial;background-repeat:init=
ial"><font face=3D"monospace, monospace">=C2=A0</font></span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%;background-image:initial;background-repeat:init=
ial"><font face=3D"monospace, monospace">Regards,</font></span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%;background-image:initial;background-repeat:init=
ial"><font face=3D"monospace, monospace">=C2=A0Rifaat</font></span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%;background-image:initial;background-repeat:init=
ial"><font face=3D"monospace, monospace">=C2=A0</font></span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%;font-family:Consolas;background-image:initial;b=
ackground-repeat:initial">=C2=A0</span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%;font-family:Consolas;background-image:initial;b=
ackground-repeat:initial">=C2=A0</span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%;font-family:Consolas;background-image:initial;b=
ackground-repeat:initial">=C2=A0</span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%;font-family:Consolas;background-image:initial;b=
ackground-repeat:initial">=C2=A0</span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%;font-family:Consolas;background-image:initial;b=
ackground-repeat:initial">=C2=A0</span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%;font-family:Consolas;background-image:initial;b=
ackground-repeat:initial">=C2=A0</span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%;font-family:Consolas">=C2=A0</span></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%;font-family:Consolas">=C2=A0</span></p></div><d=
iv class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Tue, Jun 21, 201=
6 at 7:46 PM, Peterson, Jon <span dir=3D"ltr">&lt;<a href=3D"mailto:jon.pet=
erson@neustar.biz" target=3D"_blank">jon.peterson@neustar.biz</a>&gt;</span=
> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc 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>It&#39;s not really what I&#39;m looking for, no, and after mulling th=
is over a bit, I fear going farther along this path would only drill down i=
nto levels of complexity that will likely make this harder to understand ra=
ther than easier.</div>
<div><br>
</div>
<div>I do acknowledge that you describe the INVITE as being rejected if it =
doesn&#39;t contain a token - but that&#39;s not the same thing as saying t=
he INVITE will be rejected because the relying party makes an authorization=
 decision based on the attributes in the
 token. It&#39;s more like an INVITE being rejected because a user agent ca=
n&#39;t supply an Authorization header. I will reiterate that I don&#39;t t=
hink the kinds of attributes you are proposing should be in the scope of an=
 effort that will try to address a framework
 for authorizing SIP requests. I don&#39;t think the values of the attribut=
es (as opposed to the mere presence of some attributes) have anything to do=
 with authorizing the SIP request that you propose would carry the attribut=
es.</div>
<div><br>
</div>
<div>
<div>With regards to this whole conferencing example, it does seem a bit co=
ntrived to me, like you&#39;re sketching a SIP-based way of setting confere=
nce policy on the fly that happens to require the kind of attributes you wa=
nt, and then saying that demonstrates
 the need for the attributes. The fact that all of the ways to do conferenc=
e policy we can point to in fact do it via the web makes it hard for me to =
understand why we should create a SIP-based way of doing this. To me, it lo=
oks like you&#39;re opening a problem
 space that has much more to do with reshaping the way that SIP does confer=
ences (and again, as Radhika&#39;s mail pointing out, there&#39;s been a lo=
t of thinking about this since the days of RFC4579) than it does with defin=
ing a framework for authorizing SIP requests.</div>
</div>
<div><br>
</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-BOTTOM:medium none;BORDER-LEFT:medium none;PADDING-BOTTOM:0in;PADD=
ING-LEFT:0in;PADDING-RIGHT:0in;BORDER-TOP:#b5c4df 1pt solid;BORDER-RIGHT:me=
dium 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" target=3D"_blank">rifaat.ietf@gmail.com<=
/a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Thursday, June 16, 2016 at 6:=
19 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>&quot;<a href=3D"mailto:sipcore=
@ietf.org" target=3D"_blank">sipcore@ietf.org</a>&quot; &lt;<a href=3D"mail=
to:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a>&gt;<span class=
=3D""><br>
<span style=3D"font-weight:bold">Subject: </span>Re: [sipcore] SIP AuthNZ P=
roblem Statement - v3<br>
</span></div><div><div class=3D"h5">
<div><br>
</div>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div>
<div>
<div dir=3D"ltr">
<div>The attributes I provided are examples of attributes that might be pro=
vided in an access token; it was not meant to include all the attributes fo=
r this use case, because my intention was not to focus on conference server=
, but to discuss the idea in general.</div>
<div><br>
</div>
<div>
<div>The following could be used with the conference factory to request a c=
onference focus:</div>
<div>The INVITE sent to the conference factory could include a token that h=
as a <b>
conference-type</b> attribute which could take one of the following values:=
 <b>audio, video, webconference, audio-webconference, or video-webconferenc=
e</b>.</div>
<div>The conference factory will then issue a <b>conference uri</b> for the=
 requested conference type, assuming the token is valid. If the
<b>conference-type</b> attribute is missing, then the request would be reje=
cted.</div>
</div>
<div><br>
</div>
<div>Is that what you are looking for?</div>
<div><br>
</div>
<div>Regards,</div>
<div>=C2=A0Rifaat</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Wed, Jun 15, 2016 at 5:06 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:0 0 0 .8ex;border-left:1p=
x #ccc 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><span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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">
<span><br>
<br>
</span>
<div>
<div>In that case, if you could furnish a use case with attributes that wou=
ld be carried in a SIP iNVITE that are actually salient to the authorizatio=
n of that particular SIP request, we&#39;d be getting a lot closer to commo=
n ground.</div>
<div><br>
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>An obvious example would be an INVITE with a token sent to a conferenc=
e factory to authorize the creation of a focus.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span>
<div>We&#39;re still talking past each other. My point above was that the t=
wo attributes you specified - constraints on conference size and media type=
s - have nothing to do with the authorization of the INVITE carrying those =
attributes. Like, unless the conference
 size permitted is 0, then the INVITE that creates an ad hoc conference and=
 connects you to its bridge is not going to be rejected by the focus on the=
 basis of that conference size attribute. If the maximum conference size we=
re 0, the AuthZ service just wouldn&#39;t
 attach a token to an INVITE as it would never be allowed to start a confer=
ence. Same goes for media types.</div>
<div><br>
</div>
<div>Instead, what your two attributes constrain is what happens after the =
conference is created: when ten users try to dial in to a conference with a=
 permitted size of nine, say. That tenth INVITE would get rejected - not th=
e conference-forming INVITE that
 you propose would carry the token. That&#39;s why what you&#39;re proposin=
g is overloading the INVITE to tunnel conference policy constraints to the =
focus. I mean, maybe that&#39;s a problem space worth investigating, though=
 I suspect there are good reasons why RFC4353
 specified that conference policy is not manipulated with SIP - mostly beca=
use the semantics of manipulating conference policy probably don&#39;t map =
well onto dialog-forming SIP requests. But I think that conference policy p=
rovisioning problem space would be pretty
 far from where we started in this discussion.</div>
<div><br>
</div>
<div>Let me reiterate what I wrote in my previous mail, which I quote here =
at the top: &quot;furnish a use case with attributes that would be carried =
in a SIP iNVITE that are actually salient to the authorization of that part=
icular SIP request.&quot; Again: &quot;that particular
 SIP request.&quot; Not later transactions in some broader context in which=
 the original user agent probably isn&#39;t even an endpoint.</div>
<div><br>
</div>
<div>The Authorization header in SIP, for example, is actually used by the =
UAS to make a decision about whether or not to authorize the request in whi=
ch the header appears.=C2=A0 The token that you propose a SIP INVITE would =
carry in this case is not used by the
 relying party to make a decision about whether or not that INVITE is autho=
rized. That&#39;s why I believe this conference policy issue is a different=
 problem space entirely from figuring out how to implement attribute-based =
authorization for SIP requests.</div>
<span>
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<div>Regards,</div>
<div>=C2=A0Rifaat</div>
<div><br>
</div>
</div>
</div>
</div>
</blockquote>
</span></span></div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</blockquote>
</div></div></span>
</div>

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

--001a113519241ca18d0535dd41f7--


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From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Date: Wed, 22 Jun 2016 08:48:29 -0400
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To: Roman Shpount <roman@telurix.com>
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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Roman,

I think that the question of how to exactly carry the token in SIP request
is a minor issue *at this stage*.
We are discussing the idea and concept of delegating both the AuthN and
AuthZ to separate server(s) and how to achieve that.

Regards,
 Rifaat


On Tue, Jun 21, 2016 at 7:50 PM, Roman Shpount <roman@telurix.com> wrote:

> Furthermore, reading all of this, I cannot understand why Authorization
> token cannot be a part of SIP URL, either as user part or URL parameter.
> This does not need to be anything more then service specific URL and does
> not require a new SIP header or authentication mechanism.
>
> _____________
> Roman Shpount
>
> On Tue, Jun 21, 2016 at 7:46 PM, Peterson, Jon <jon.peterson@neustar.biz>
> wrote:
>
>>
>> It's not really what I'm looking for, no, and after mulling this over a
>> bit, I fear going farther along this path would only drill down into levels
>> of complexity that will likely make this harder to understand rather than
>> easier.
>>
>> I do acknowledge that you describe the INVITE as being rejected if it
>> doesn't contain a token - but that's not the same thing as saying the
>> INVITE will be rejected because the relying party makes an authorization
>> decision based on the attributes in the token. It's more like an INVITE
>> being rejected because a user agent can't supply an Authorization header. I
>> will reiterate that I don't think the kinds of attributes you are proposing
>> should be in the scope of an effort that will try to address a framework
>> for authorizing SIP requests. I don't think the values of the attributes
>> (as opposed to the mere presence of some attributes) have anything to do
>> with authorizing the SIP request that you propose would carry the
>> attributes.
>>
>> With regards to this whole conferencing example, it does seem a bit
>> contrived to me, like you're sketching a SIP-based way of setting
>> conference policy on the fly that happens to require the kind of attributes
>> you want, and then saying that demonstrates the need for the attributes.
>> The fact that all of the ways to do conference policy we can point to in
>> fact do it via the web makes it hard for me to understand why we should
>> create a SIP-based way of doing this. To me, it looks like you're opening a
>> problem space that has much more to do with reshaping the way that SIP does
>> conferences (and again, as Radhika's mail pointing out, there's been a lot
>> of thinking about this since the days of RFC4579) than it does with
>> defining a framework for authorizing SIP requests.
>>
>> Jon Peterson
>> Neustar, Inc.
>>
>> From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
>> Date: Thursday, June 16, 2016 at 6:19 AM
>> To: Jon Peterson <jon.peterson@neustar.biz>
>> Cc: "sipcore@ietf.org" <sipcore@ietf.org>
>> Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
>>
>> The attributes I provided are examples of attributes that might be
>> provided in an access token; it was not meant to include all the attributes
>> for this use case, because my intention was not to focus on conference
>> server, but to discuss the idea in general.
>>
>> The following could be used with the conference factory to request a
>> conference focus:
>> The INVITE sent to the conference factory could include a token that has
>> a * conference-type* attribute which could take one of the following
>> values: *audio, video, webconference, audio-webconference, or
>> video-webconference*.
>> The conference factory will then issue a *conference uri* for the
>> requested conference type, assuming the token is valid. If the
>> *conference-type* attribute is missing, then the request would be
>> rejected.
>>
>> Is that what you are looking for?
>>
>> Regards,
>>  Rifaat
>>
>>
>>
>>
>>
>> On Wed, Jun 15, 2016 at 5:06 PM, Peterson, Jon <jon.peterson@neustar.biz>
>> wrote:
>>
>>>
>>>>
>>>> In that case, if you could furnish a use case with attributes that
>>>> would be carried in a SIP iNVITE that are actually salient to the
>>>> authorization of that particular SIP request, we'd be getting a lot closer
>>>> to common ground.
>>>>
>>>>
>>> An obvious example would be an INVITE with a token sent to a conference
>>> factory to authorize the creation of a focus.
>>>
>>>
>>> We're still talking past each other. My point above was that the two
>>> attributes you specified - constraints on conference size and media types -
>>> have nothing to do with the authorization of the INVITE carrying those
>>> attributes. Like, unless the conference size permitted is 0, then the
>>> INVITE that creates an ad hoc conference and connects you to its bridge is
>>> not going to be rejected by the focus on the basis of that conference size
>>> attribute. If the maximum conference size were 0, the AuthZ service just
>>> wouldn't attach a token to an INVITE as it would never be allowed to start
>>> a conference. Same goes for media types.
>>>
>>> Instead, what your two attributes constrain is what happens after the
>>> conference is created: when ten users try to dial in to a conference with a
>>> permitted size of nine, say. That tenth INVITE would get rejected - not the
>>> conference-forming INVITE that you propose would carry the token. That's
>>> why what you're proposing is overloading the INVITE to tunnel conference
>>> policy constraints to the focus. I mean, maybe that's a problem space worth
>>> investigating, though I suspect there are good reasons why RFC4353
>>> specified that conference policy is not manipulated with SIP - mostly
>>> because the semantics of manipulating conference policy probably don't map
>>> well onto dialog-forming SIP requests. But I think that conference policy
>>> provisioning problem space would be pretty far from where we started in
>>> this discussion.
>>>
>>> Let me reiterate what I wrote in my previous mail, which I quote here at
>>> the top: "furnish a use case with attributes that would be carried in a SIP
>>> iNVITE that are actually salient to the authorization of that particular
>>> SIP request." Again: "that particular SIP request." Not later transactions
>>> in some broader context in which the original user agent probably isn't
>>> even an endpoint.
>>>
>>> The Authorization header in SIP, for example, is actually used by the
>>> UAS to make a decision about whether or not to authorize the request in
>>> which the header appears.  The token that you propose a SIP INVITE would
>>> carry in this case is not used by the relying party to make a decision
>>> about whether or not that INVITE is authorized. That's why I believe this
>>> conference policy issue is a different problem space entirely from figuring
>>> out how to implement attribute-based authorization for SIP requests.
>>>
>>> Jon Peterson
>>> Neustar, Inc.
>>>
>>>
>>> Regards,
>>>  Rifaat
>>>
>>>
>>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
>>
>

--001a113ddfbccc5c550535dd5a56
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Roman,<div><br></div><div>I think that the question of how=
 to exactly carry the token in SIP request is a minor issue <b>at this stag=
e</b>.<br></div><div>We are discussing the idea and concept of delegating b=
oth the AuthN and AuthZ to separate server(s) and how to achieve that.<br><=
/div><div><br></div><div>Regards,</div><div>=C2=A0Rifaat</div><div><br></di=
v></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Tue, J=
un 21, 2016 at 7:50 PM, Roman Shpount <span dir=3D"ltr">&lt;<a href=3D"mail=
to:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt;</span> wr=
ote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border=
-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr">Furthermore, readin=
g all of this, I cannot understand why Authorization token cannot be a part=
 of SIP URL, either as user part or URL parameter. This does not need to be=
 anything more then service specific URL and does not require a new SIP hea=
der or authentication mechanism.</div><div class=3D"gmail_extra"><br clear=
=3D"all"><div><div data-smartmail=3D"gmail_signature">_____________<br>Roma=
n Shpount</div></div>
<br><div class=3D"gmail_quote"><div><div class=3D"h5">On Tue, Jun 21, 2016 =
at 7:46 PM, Peterson, Jon <span dir=3D"ltr">&lt;<a href=3D"mailto:jon.peter=
son@neustar.biz" target=3D"_blank">jon.peterson@neustar.biz</a>&gt;</span> =
wrote:<br></div></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0=
 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class=3D"h5"=
>



<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>It&#39;s not really what I&#39;m looking for, no, and after mulling th=
is over a bit, I fear going farther along this path would only drill down i=
nto levels of complexity that will likely make this harder to understand ra=
ther than easier.</div>
<div><br>
</div>
<div>I do acknowledge that you describe the INVITE as being rejected if it =
doesn&#39;t contain a token - but that&#39;s not the same thing as saying t=
he INVITE will be rejected because the relying party makes an authorization=
 decision based on the attributes in the
 token. It&#39;s more like an INVITE being rejected because a user agent ca=
n&#39;t supply an Authorization header. I will reiterate that I don&#39;t t=
hink the kinds of attributes you are proposing should be in the scope of an=
 effort that will try to address a framework
 for authorizing SIP requests. I don&#39;t think the values of the attribut=
es (as opposed to the mere presence of some attributes) have anything to do=
 with authorizing the SIP request that you propose would carry the attribut=
es.</div>
<div><br>
</div>
<div>
<div>With regards to this whole conferencing example, it does seem a bit co=
ntrived to me, like you&#39;re sketching a SIP-based way of setting confere=
nce policy on the fly that happens to require the kind of attributes you wa=
nt, and then saying that demonstrates
 the need for the attributes. The fact that all of the ways to do conferenc=
e policy we can point to in fact do it via the web makes it hard for me to =
understand why we should create a SIP-based way of doing this. To me, it lo=
oks like you&#39;re opening a problem
 space that has much more to do with reshaping the way that SIP does confer=
ences (and again, as Radhika&#39;s mail pointing out, there&#39;s been a lo=
t of thinking about this since the days of RFC4579) than it does with defin=
ing a framework for authorizing SIP requests.</div>
</div>
<div><br>
</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-BOTTOM:medium none;BORDER-LEFT:medium none;PADDING-BOTTOM:0in;PADD=
ING-LEFT:0in;PADDING-RIGHT:0in;BORDER-TOP:#b5c4df 1pt solid;BORDER-RIGHT:me=
dium 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" target=3D"_blank">rifaat.ietf@gmail.com<=
/a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Thursday, June 16, 2016 at 6:=
19 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>&quot;<a href=3D"mailto:sipcore=
@ietf.org" target=3D"_blank">sipcore@ietf.org</a>&quot; &lt;<a href=3D"mail=
to:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [sipcore] SIP AuthNZ P=
roblem Statement - v3<br>
</div>
<div><br>
</div>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div>
<div>
<div dir=3D"ltr">
<div>The attributes I provided are examples of attributes that might be pro=
vided in an access token; it was not meant to include all the attributes fo=
r this use case, because my intention was not to focus on conference server=
, but to discuss the idea in general.</div>
<div><br>
</div>
<div>
<div>The following could be used with the conference factory to request a c=
onference focus:</div>
<div>The INVITE sent to the conference factory could include a token that h=
as a <b>
conference-type</b> attribute which could take one of the following values:=
 <b>audio, video, webconference, audio-webconference, or video-webconferenc=
e</b>.</div>
<div>The conference factory will then issue a <b>conference uri</b> for the=
 requested conference type, assuming the token is valid. If the
<b>conference-type</b> attribute is missing, then the request would be reje=
cted.</div>
</div>
<div><br>
</div>
<div>Is that what you are looking for?</div>
<div><br>
</div>
<div>Regards,</div>
<div>=C2=A0Rifaat</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Wed, Jun 15, 2016 at 5:06 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:0 0 0 .8ex;border-left:1p=
x #ccc 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><span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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">
<span><br>
<br>
</span>
<div>
<div>In that case, if you could furnish a use case with attributes that wou=
ld be carried in a SIP iNVITE that are actually salient to the authorizatio=
n of that particular SIP request, we&#39;d be getting a lot closer to commo=
n ground.</div>
<div><br>
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>An obvious example would be an INVITE with a token sent to a conferenc=
e factory to authorize the creation of a focus.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span>
<div>We&#39;re still talking past each other. My point above was that the t=
wo attributes you specified - constraints on conference size and media type=
s - have nothing to do with the authorization of the INVITE carrying those =
attributes. Like, unless the conference
 size permitted is 0, then the INVITE that creates an ad hoc conference and=
 connects you to its bridge is not going to be rejected by the focus on the=
 basis of that conference size attribute. If the maximum conference size we=
re 0, the AuthZ service just wouldn&#39;t
 attach a token to an INVITE as it would never be allowed to start a confer=
ence. Same goes for media types.</div>
<div><br>
</div>
<div>Instead, what your two attributes constrain is what happens after the =
conference is created: when ten users try to dial in to a conference with a=
 permitted size of nine, say. That tenth INVITE would get rejected - not th=
e conference-forming INVITE that
 you propose would carry the token. That&#39;s why what you&#39;re proposin=
g is overloading the INVITE to tunnel conference policy constraints to the =
focus. I mean, maybe that&#39;s a problem space worth investigating, though=
 I suspect there are good reasons why RFC4353
 specified that conference policy is not manipulated with SIP - mostly beca=
use the semantics of manipulating conference policy probably don&#39;t map =
well onto dialog-forming SIP requests. But I think that conference policy p=
rovisioning problem space would be pretty
 far from where we started in this discussion.</div>
<div><br>
</div>
<div>Let me reiterate what I wrote in my previous mail, which I quote here =
at the top: &quot;furnish a use case with attributes that would be carried =
in a SIP iNVITE that are actually salient to the authorization of that part=
icular SIP request.&quot; Again: &quot;that particular
 SIP request.&quot; Not later transactions in some broader context in which=
 the original user agent probably isn&#39;t even an endpoint.</div>
<div><br>
</div>
<div>The Authorization header in SIP, for example, is actually used by the =
UAS to make a decision about whether or not to authorize the request in whi=
ch the header appears.=C2=A0 The token that you propose a SIP INVITE would =
carry in this case is not used by the
 relying party to make a decision about whether or not that INVITE is autho=
rized. That&#39;s why I believe this conference policy issue is a different=
 problem space entirely from figuring out how to implement attribute-based =
authorization for SIP requests.</div>
<span>
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<div>Regards,</div>
<div>=C2=A0Rifaat</div>
<div><br>
</div>
</div>
</div>
</div>
</blockquote>
</span></span></div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</blockquote>
</span>
</div>

<br></div></div><span class=3D"">__________________________________________=
_____<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a><=
br>
</span><a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" rel=3D"nor=
eferrer" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sipcore</a=
><br>
<br></blockquote></div><br></div>
</blockquote></div><br></div>

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Date: Wed, 22 Jun 2016 09:47:05 -0500
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Subject: Re: [sipcore] AD Evaluation of draft-holmberg-dispatch-rfc7315-updates-06
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Works for me.

Thanks!

Ben.

On 22 Jun 2016, at 2:18, Christer Holmberg wrote:

> Hi,
>
> NEW:
>
>    =E2=80=9DThe security considerations for these P- header fields are =

> defined in
>    [RFC7315].  This specification allows some header fields to be
>    present in messages where they were previously not allowed, and the
>    security considerations and assumptions (e.g. regarding only =

> sending
>    Information to trusted entities) also to those messages. In =

> addition,
>    this specification also disallow some header fields to be present
>    in message where they were previously allowed. That does not cause
>    any security issues, but implementations need to be aware that
>    implementations may not have been updated according to this =

> document,
>    and take proper actions if a header field occur, or does not occur,
>    in a message where it should occur (or occurs in a message where it
>    should not occur). This document adds the ability to include
>    P-Access-Network-Info in ACK requests. As documented in  [RFC7315],
>    P-Access-Network-Info may include privacy sensitive information, =

> including
>    the user's location. The security and privacy considerations for
>    P-Access-Network-Info in ACK requests are similar to those for the =

> other
>    SIP requests discussed in  [RFC7315].=E2=80=9D
>
> Regards,
>
> Christer
>
>
> From: Christer Holmberg =

> <christer.holmberg@ericsson.com<mailto:christer.holmberg@ericsson.com>>=

> Date: Wednesday 22 June 2016 at 05:28
> To: "gsalguei@cisco.com<mailto:gsalguei@cisco.com>" =

> <gsalguei@cisco.com<mailto:gsalguei@cisco.com>>, Ben Campbell =

> <ben@nostrum.com<mailto:ben@nostrum.com>>
> Cc: =

> "draft-holmberg-dispatch-rfc7315-updates.all@ietf.org<mailto:draft-holm=
berg-dispatch-rfc7315-updates.all@ietf.org>" =

> <draft-holmberg-dispatch-rfc7315-updates.all@ietf.org<mailto:draft-holm=
berg-dispatch-rfc7315-updates.all@ietf.org>>, =

> "sipcore@ietf.org<mailto:sipcore@ietf.org>" =

> <sipcore@ietf.org<mailto:sipcore@ietf.org>>
> Subject: RE: AD Evaluation of =

> draft-holmberg-dispatch-rfc7315-updates-06
> Resent-From: <alias-bounces@ietf.org<mailto:alias-bounces@ietf.org>>
> Resent-To: Christer Holmberg =

> <christer.holmberg@ericsson.com<mailto:christer.holmberg@ericsson.com>>=
, =

> Nevenka Biondic =

> <nevenka.biondic@ericsson.com<mailto:nevenka.biondic@ericsson.com>>, =

> "gsalguei@cisco.com<mailto:gsalguei@cisco.com>" =

> <gsalguei@cisco.com<mailto:gsalguei@cisco.com>>, Ben Campbell =

> <ben@nostrum.com<mailto:ben@nostrum.com>>, "A. Mahoney" =

> <mahoney@nostrum.com<mailto:mahoney@nostrum.com>>
> Resent-Date: Wednesday 22 June 2016 at 05:28
>
> Hi,
>
> We can add the text.
>
> Regards,
>
> Christer
>
> Sent from my Windows Phone
> ________________________________
> From: Gonzalo Salgueiro (gsalguei)<mailto:gsalguei@cisco.com>
> Sent: 21/06/2016 19:44
> To: Ben Campbell<mailto:ben@nostrum.com>
> Cc: Christer Holmberg<mailto:christer.holmberg@ericsson.com>; =

> draft-holmberg-dispatch-rfc7315-updates.all@ietf.org<mailto:draft-holmb=
erg-dispatch-rfc7315-updates.all@ietf.org>; =

> SIPCORE<mailto:sipcore@ietf.org>
> Subject: Re: AD Evaluation of =

> draft-holmberg-dispatch-rfc7315-updates-06
>
>
>> On Jun 21, 2016, at 11:22 AM, Ben Campbell =

>> <ben@nostrum.com<mailto:ben@nostrum.com>> wrote:
>>
>> That's a good start, but don't be surprised if we get questions =

>> specifically about adding NPLI to ACK requests. some language to the =

>> effect of the following might help:
>>
>> "This document adds the ability to include P-Access-Network-Info in =

>> ACK requests. As documented in RFC7315, P-Access-Network-Info may =

>> include privacy sensitive information, including the user's location. =

>> The security and privacy considerations for P-Access-Network-Info in =

>> ACK requests are similar to those for the other SIP requests =

>> discussed in RFC7315.=E2=80=9D
>
> I=E2=80=99m fine with adding such text.
>
> Christer - Can we append this to your proposed text?
>
> Gonzalo
>
>
>>
>> Thanks!
>>
>> Ben.
>>
>> On 21 Jun 2016, at 3:26, Christer Holmberg wrote:
>>
>>> Hi Ben,
>>>
>>> See inline.
>>>
>>>> --------------
>>>>
>>>> Substantive:
>>>>
>>>> The security considerations state that the draft removes some =

>>>> places
>>>> that some of the P-Headers can be sent, but expands that to some =

>>>> other
>>>> places. Further, it says that neither introduce new security
>>>> considerations beyond those in 7315.
>>>>
>>>> I accept that for the reduction part. But I'm not sure we can state =

>>>> that
>>>> sort of thing for the expansion part, at least without some more
>>>> discussion. Since 7315 already acknowledges potential privacy =

>>>> issues
>>>> around P-Access-Network-Info, I'd like to at least see a sentence =

>>>> or two
>>>> about the allowance of that in ACK requests, even if they just say =

>>>> that
>>>> this addition makes things no worse than they already are.
>>>
>>>
>>> OLD:
>>>
>>> The security considerations for P- header fields are defined in
>>>   [RFC7315].  This specification allows some header fields to be
>>>   present in messages where they were previously not allowed, and
>>>   disallow some header fields to be present in messages where they =

>>> were
>>>   previously allowed. That does not cause any security issues, but
>>>   implementations need to be aware that implementations may not have
>>>   been updated according to this document, and take proper actions =

>>> if a
>>>   header field occur, or does not occur, in a message where it =

>>> should
>>>   occur (or occurs in a message where it should not occur).
>>>
>>>
>>>
>>> NEW:
>>>
>>> The security considerations for these P- header fields are defined =

>>> in
>>>   [RFC7315].  This specification allows some header fields to be
>>>   present in messages where they were previously not allowed, and =

>>> the
>>> security considerations and assumptions (e.g. regarding only sending
>>> Information to trusted entities) also to those messages. In =

>>> addition,
>>> this specification also disallow some header fields to be present
>>> in message where they were previously allowed. That does not cause
>>> any security issues, but implementations need to be aware that
>>> implementations may not have been updated according to this =

>>> document,
>>> and take proper actions if a header field occur, or does not occur,
>>> in a message where it should occur (or occurs in a message where it
>>> should not occur).
>>>
>>>
>>>
>>>> Editorial:
>>>>
>>>> -5, first sentence: "The security considerations for P- header =

>>>> fields
>>>> are defined in
>>>>   [RFC7315]"
>>>> I assume this means 7315 discusses the security considerations for =

>>>> these
>>>> P-Headers specifically, not P-Headers in general. Is this the =

>>>> intent? If
>>>> so, I suggest:
>>>>
>>>> s/... for P-header fields.../ ... for these P-header fields...
>>>
>>> I=C2=B9ll fix as suggested (ass new text above).
>>>
>>> Regards,
>>>
>>> Christer


From nobody Wed Jun 22 08:11:20 2016
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Ben Campbell <ben@nostrum.com>
Thread-Topic: AD Evaluation of draft-holmberg-dispatch-rfc7315-updates-06
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References: <87A3DCDE-B8BC-4ADE-8129-70A4C0E92C3D@nostrum.com> <D38ED131.B2A5%christer.holmberg@ericsson.com> <54648860-7461-4A4E-948A-A1C9FAAC7FFC@nostrum.com> <83801023-F21E-417C-B49C-49820CCE4DF2@cisco.com> <7594FB04B1934943A5C02806D1A2204B380FB854@ESESSMB209.ericsson.se> <D3901671.B451%christer.holmberg@ericsson.com>, <8D74E280-1141-469D-9627-23E38A2F9478@nostrum.com>
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Subject: Re: [sipcore] AD Evaluation of draft-holmberg-dispatch-rfc7315-updates-06
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Good :)

Is it ok of I submit a new version of the draft? That way the new text will=
 be available during  IETF last call.

Regards,

Christer

Sent from my Windows Phone
________________________________
From: Ben Campbell<mailto:ben@nostrum.com>
Sent: =FD22/=FD06/=FD2016 17:47
To: Christer Holmberg<mailto:christer.holmberg@ericsson.com>
Cc: Gonzalo Salgueiro<mailto:gsalguei@cisco.com>; draft-holmberg-dispatch-r=
fc7315-updates.all@ietf.org<mailto:draft-holmberg-dispatch-rfc7315-updates.=
all@ietf.org>; SIPCORE<mailto:sipcore@ietf.org>
Subject: Re: AD Evaluation of draft-holmberg-dispatch-rfc7315-updates-06

Works for me.

Thanks!

Ben.

On 22 Jun 2016, at 2:18, Christer Holmberg wrote:

> Hi,
>
> NEW:
>
>    =94The security considerations for these P- header fields are
> defined in
>    [RFC7315].  This specification allows some header fields to be
>    present in messages where they were previously not allowed, and the
>    security considerations and assumptions (e.g. regarding only
> sending
>    Information to trusted entities) also to those messages. In
> addition,
>    this specification also disallow some header fields to be present
>    in message where they were previously allowed. That does not cause
>    any security issues, but implementations need to be aware that
>    implementations may not have been updated according to this
> document,
>    and take proper actions if a header field occur, or does not occur,
>    in a message where it should occur (or occurs in a message where it
>    should not occur). This document adds the ability to include
>    P-Access-Network-Info in ACK requests. As documented in  [RFC7315],
>    P-Access-Network-Info may include privacy sensitive information,
> including
>    the user's location. The security and privacy considerations for
>    P-Access-Network-Info in ACK requests are similar to those for the
> other
>    SIP requests discussed in  [RFC7315].=94
>
> Regards,
>
> Christer
>
>
> From: Christer Holmberg
> <christer.holmberg@ericsson.com<mailto:christer.holmberg@ericsson.com>>
> Date: Wednesday 22 June 2016 at 05:28
> To: "gsalguei@cisco.com<mailto:gsalguei@cisco.com>"
> <gsalguei@cisco.com<mailto:gsalguei@cisco.com>>, Ben Campbell
> <ben@nostrum.com<mailto:ben@nostrum.com>>
> Cc:
> "draft-holmberg-dispatch-rfc7315-updates.all@ietf.org<mailto:draft-holmbe=
rg-dispatch-rfc7315-updates.all@ietf.org>"
> <draft-holmberg-dispatch-rfc7315-updates.all@ietf.org<mailto:draft-holmbe=
rg-dispatch-rfc7315-updates.all@ietf.org>>,
> "sipcore@ietf.org<mailto:sipcore@ietf.org>"
> <sipcore@ietf.org<mailto:sipcore@ietf.org>>
> Subject: RE: AD Evaluation of
> draft-holmberg-dispatch-rfc7315-updates-06
> Resent-From: <alias-bounces@ietf.org<mailto:alias-bounces@ietf.org>>
> Resent-To: Christer Holmberg
> <christer.holmberg@ericsson.com<mailto:christer.holmberg@ericsson.com>>,
> Nevenka Biondic
> <nevenka.biondic@ericsson.com<mailto:nevenka.biondic@ericsson.com>>,
> "gsalguei@cisco.com<mailto:gsalguei@cisco.com>"
> <gsalguei@cisco.com<mailto:gsalguei@cisco.com>>, Ben Campbell
> <ben@nostrum.com<mailto:ben@nostrum.com>>, "A. Mahoney"
> <mahoney@nostrum.com<mailto:mahoney@nostrum.com>>
> Resent-Date: Wednesday 22 June 2016 at 05:28
>
> Hi,
>
> We can add the text.
>
> Regards,
>
> Christer
>
> Sent from my Windows Phone
> ________________________________
> From: Gonzalo Salgueiro (gsalguei)<mailto:gsalguei@cisco.com>
> Sent: 21/06/2016 19:44
> To: Ben Campbell<mailto:ben@nostrum.com>
> Cc: Christer Holmberg<mailto:christer.holmberg@ericsson.com>;
> draft-holmberg-dispatch-rfc7315-updates.all@ietf.org<mailto:draft-holmber=
g-dispatch-rfc7315-updates.all@ietf.org>;
> SIPCORE<mailto:sipcore@ietf.org>
> Subject: Re: AD Evaluation of
> draft-holmberg-dispatch-rfc7315-updates-06
>
>
>> On Jun 21, 2016, at 11:22 AM, Ben Campbell
>> <ben@nostrum.com<mailto:ben@nostrum.com>> wrote:
>>
>> That's a good start, but don't be surprised if we get questions
>> specifically about adding NPLI to ACK requests. some language to the
>> effect of the following might help:
>>
>> "This document adds the ability to include P-Access-Network-Info in
>> ACK requests. As documented in RFC7315, P-Access-Network-Info may
>> include privacy sensitive information, including the user's location.
>> The security and privacy considerations for P-Access-Network-Info in
>> ACK requests are similar to those for the other SIP requests
>> discussed in RFC7315.=94
>
> I=92m fine with adding such text.
>
> Christer - Can we append this to your proposed text?
>
> Gonzalo
>
>
>>
>> Thanks!
>>
>> Ben.
>>
>> On 21 Jun 2016, at 3:26, Christer Holmberg wrote:
>>
>>> Hi Ben,
>>>
>>> See inline.
>>>
>>>> --------------
>>>>
>>>> Substantive:
>>>>
>>>> The security considerations state that the draft removes some
>>>> places
>>>> that some of the P-Headers can be sent, but expands that to some
>>>> other
>>>> places. Further, it says that neither introduce new security
>>>> considerations beyond those in 7315.
>>>>
>>>> I accept that for the reduction part. But I'm not sure we can state
>>>> that
>>>> sort of thing for the expansion part, at least without some more
>>>> discussion. Since 7315 already acknowledges potential privacy
>>>> issues
>>>> around P-Access-Network-Info, I'd like to at least see a sentence
>>>> or two
>>>> about the allowance of that in ACK requests, even if they just say
>>>> that
>>>> this addition makes things no worse than they already are.
>>>
>>>
>>> OLD:
>>>
>>> The security considerations for P- header fields are defined in
>>>   [RFC7315].  This specification allows some header fields to be
>>>   present in messages where they were previously not allowed, and
>>>   disallow some header fields to be present in messages where they
>>> were
>>>   previously allowed. That does not cause any security issues, but
>>>   implementations need to be aware that implementations may not have
>>>   been updated according to this document, and take proper actions
>>> if a
>>>   header field occur, or does not occur, in a message where it
>>> should
>>>   occur (or occurs in a message where it should not occur).
>>>
>>>
>>>
>>> NEW:
>>>
>>> The security considerations for these P- header fields are defined
>>> in
>>>   [RFC7315].  This specification allows some header fields to be
>>>   present in messages where they were previously not allowed, and
>>> the
>>> security considerations and assumptions (e.g. regarding only sending
>>> Information to trusted entities) also to those messages. In
>>> addition,
>>> this specification also disallow some header fields to be present
>>> in message where they were previously allowed. That does not cause
>>> any security issues, but implementations need to be aware that
>>> implementations may not have been updated according to this
>>> document,
>>> and take proper actions if a header field occur, or does not occur,
>>> in a message where it should occur (or occurs in a message where it
>>> should not occur).
>>>
>>>
>>>
>>>> Editorial:
>>>>
>>>> -5, first sentence: "The security considerations for P- header
>>>> fields
>>>> are defined in
>>>>   [RFC7315]"
>>>> I assume this means 7315 discusses the security considerations for
>>>> these
>>>> P-Headers specifically, not P-Headers in general. Is this the
>>>> intent? If
>>>> so, I suggest:
>>>>
>>>> s/... for P-header fields.../ ... for these P-header fields...
>>>
>>> I=B9ll fix as suggested (ass new text above).
>>>
>>> Regards,
>>>
>>> Christer

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<div>
<div>
<div style=3D"font-family:Calibri,sans-serif; font-size:11pt">Good :)<br>
<br>
Is it ok of I submit a new version of the draft? That way the new text will=
 be available during&nbsp; IETF last call.<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:ben@nostrum.com">Ben Campbell</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">=FD22=
/=FD06/=FD2016 17:47</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:christer.holmberg@ericsson.com">Christer Holmberg</a></span><b=
r>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Cc:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:gsalguei@cisco.com">Gonzalo Salgueiro</a>;
<a href=3D"mailto:draft-holmberg-dispatch-rfc7315-updates.all@ietf.org">dra=
ft-holmberg-dispatch-rfc7315-updates.all@ietf.org</a>;
<a href=3D"mailto:sipcore@ietf.org">SIPCORE</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: A=
D Evaluation of draft-holmberg-dispatch-rfc7315-updates-06</span><br>
<br>
</div>
</div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">Works for me.<br>
<br>
Thanks!<br>
<br>
Ben.<br>
<br>
On 22 Jun 2016, at 2:18, Christer Holmberg wrote:<br>
<br>
&gt; Hi,<br>
&gt;<br>
&gt; NEW:<br>
&gt;<br>
&gt;&nbsp;&nbsp;&nbsp; =94The security considerations for these P- header f=
ields are <br>
&gt; defined in<br>
&gt;&nbsp;&nbsp;&nbsp; [RFC7315].&nbsp; This specification allows some head=
er fields to be<br>
&gt;&nbsp;&nbsp;&nbsp; present in messages where they were previously not a=
llowed, and the<br>
&gt;&nbsp;&nbsp;&nbsp; security considerations and assumptions (e.g. regard=
ing only <br>
&gt; sending<br>
&gt;&nbsp;&nbsp;&nbsp; Information to trusted entities) also to those messa=
ges. In <br>
&gt; addition,<br>
&gt;&nbsp;&nbsp;&nbsp; this specification also disallow some header fields =
to be present<br>
&gt;&nbsp;&nbsp;&nbsp; in message where they were previously allowed. That =
does not cause<br>
&gt;&nbsp;&nbsp;&nbsp; any security issues, but implementations need to be =
aware that<br>
&gt;&nbsp;&nbsp;&nbsp; implementations may not have been updated according =
to this <br>
&gt; document,<br>
&gt;&nbsp;&nbsp;&nbsp; and take proper actions if a header field occur, or =
does not occur,<br>
&gt;&nbsp;&nbsp;&nbsp; in a message where it should occur (or occurs in a m=
essage where it<br>
&gt;&nbsp;&nbsp;&nbsp; should not occur). This document adds the ability to=
 include<br>
&gt;&nbsp;&nbsp;&nbsp; P-Access-Network-Info in ACK requests. As documented=
 in&nbsp; [RFC7315],<br>
&gt;&nbsp;&nbsp;&nbsp; P-Access-Network-Info may include privacy sensitive =
information, <br>
&gt; including<br>
&gt;&nbsp;&nbsp;&nbsp; the user's location. The security and privacy consid=
erations for<br>
&gt;&nbsp;&nbsp;&nbsp; P-Access-Network-Info in ACK requests are similar to=
 those for the <br>
&gt; other<br>
&gt;&nbsp;&nbsp;&nbsp; SIP requests discussed in&nbsp; [RFC7315].=94<br>
&gt;<br>
&gt; Regards,<br>
&gt;<br>
&gt; Christer<br>
&gt;<br>
&gt;<br>
&gt; From: Christer Holmberg <br>
&gt; &lt;christer.holmberg@ericsson.com&lt;mailto:christer.holmberg@ericsso=
n.com&gt;&gt;<br>
&gt; Date: Wednesday 22 June 2016 at 05:28<br>
&gt; To: &quot;gsalguei@cisco.com&lt;mailto:gsalguei@cisco.com&gt;&quot; <b=
r>
&gt; &lt;gsalguei@cisco.com&lt;mailto:gsalguei@cisco.com&gt;&gt;, Ben Campb=
ell <br>
&gt; &lt;ben@nostrum.com&lt;mailto:ben@nostrum.com&gt;&gt;<br>
&gt; Cc: <br>
&gt; &quot;draft-holmberg-dispatch-rfc7315-updates.all@ietf.org&lt;mailto:d=
raft-holmberg-dispatch-rfc7315-updates.all@ietf.org&gt;&quot;
<br>
&gt; &lt;draft-holmberg-dispatch-rfc7315-updates.all@ietf.org&lt;mailto:dra=
ft-holmberg-dispatch-rfc7315-updates.all@ietf.org&gt;&gt;,
<br>
&gt; &quot;sipcore@ietf.org&lt;mailto:sipcore@ietf.org&gt;&quot; <br>
&gt; &lt;sipcore@ietf.org&lt;mailto:sipcore@ietf.org&gt;&gt;<br>
&gt; Subject: RE: AD Evaluation of <br>
&gt; draft-holmberg-dispatch-rfc7315-updates-06<br>
&gt; Resent-From: &lt;alias-bounces@ietf.org&lt;mailto:alias-bounces@ietf.o=
rg&gt;&gt;<br>
&gt; Resent-To: Christer Holmberg <br>
&gt; &lt;christer.holmberg@ericsson.com&lt;mailto:christer.holmberg@ericsso=
n.com&gt;&gt;, <br>
&gt; Nevenka Biondic <br>
&gt; &lt;nevenka.biondic@ericsson.com&lt;mailto:nevenka.biondic@ericsson.co=
m&gt;&gt;, <br>
&gt; &quot;gsalguei@cisco.com&lt;mailto:gsalguei@cisco.com&gt;&quot; <br>
&gt; &lt;gsalguei@cisco.com&lt;mailto:gsalguei@cisco.com&gt;&gt;, Ben Campb=
ell <br>
&gt; &lt;ben@nostrum.com&lt;mailto:ben@nostrum.com&gt;&gt;, &quot;A. Mahone=
y&quot; <br>
&gt; &lt;mahoney@nostrum.com&lt;mailto:mahoney@nostrum.com&gt;&gt;<br>
&gt; Resent-Date: Wednesday 22 June 2016 at 05:28<br>
&gt;<br>
&gt; Hi,<br>
&gt;<br>
&gt; We can add the text.<br>
&gt;<br>
&gt; Regards,<br>
&gt;<br>
&gt; Christer<br>
&gt;<br>
&gt; Sent from my Windows Phone<br>
&gt; ________________________________<br>
&gt; From: Gonzalo Salgueiro (gsalguei)&lt;mailto:gsalguei@cisco.com&gt;<br=
>
&gt; Sent: 21/06/2016 19:44<br>
&gt; To: Ben Campbell&lt;<a href=3D"mailto:ben@nostrum.com">mailto:ben@nost=
rum.com</a>&gt;<br>
&gt; Cc: Christer Holmberg&lt;<a href=3D"mailto:christer.holmberg@ericsson.=
com">mailto:christer.holmberg@ericsson.com</a>&gt;;
<br>
&gt; draft-holmberg-dispatch-rfc7315-updates.all@ietf.org&lt;mailto:draft-h=
olmberg-dispatch-rfc7315-updates.all@ietf.org&gt;;
<br>
&gt; SIPCORE&lt;<a href=3D"mailto:sipcore@ietf.org">mailto:sipcore@ietf.org=
</a>&gt;<br>
&gt; Subject: Re: AD Evaluation of <br>
&gt; draft-holmberg-dispatch-rfc7315-updates-06<br>
&gt;<br>
&gt;<br>
&gt;&gt; On Jun 21, 2016, at 11:22 AM, Ben Campbell <br>
&gt;&gt; &lt;ben@nostrum.com&lt;mailto:ben@nostrum.com&gt;&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt; That's a good start, but don't be surprised if we get questions <b=
r>
&gt;&gt; specifically about adding NPLI to ACK requests. some language to t=
he <br>
&gt;&gt; effect of the following might help:<br>
&gt;&gt;<br>
&gt;&gt; &quot;This document adds the ability to include P-Access-Network-I=
nfo in <br>
&gt;&gt; ACK requests. As documented in RFC7315, P-Access-Network-Info may =
<br>
&gt;&gt; include privacy sensitive information, including the user's locati=
on. <br>
&gt;&gt; The security and privacy considerations for P-Access-Network-Info =
in <br>
&gt;&gt; ACK requests are similar to those for the other SIP requests <br>
&gt;&gt; discussed in RFC7315.=94<br>
&gt;<br>
&gt; I=92m fine with adding such text.<br>
&gt;<br>
&gt; Christer - Can we append this to your proposed text?<br>
&gt;<br>
&gt; Gonzalo<br>
&gt;<br>
&gt;<br>
&gt;&gt;<br>
&gt;&gt; Thanks!<br>
&gt;&gt;<br>
&gt;&gt; Ben.<br>
&gt;&gt;<br>
&gt;&gt; On 21 Jun 2016, at 3:26, Christer Holmberg wrote:<br>
&gt;&gt;<br>
&gt;&gt;&gt; Hi Ben,<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; See inline.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; --------------<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; Substantive:<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; The security considerations state that the draft removes s=
ome <br>
&gt;&gt;&gt;&gt; places<br>
&gt;&gt;&gt;&gt; that some of the P-Headers can be sent, but expands that t=
o some <br>
&gt;&gt;&gt;&gt; other<br>
&gt;&gt;&gt;&gt; places. Further, it says that neither introduce new securi=
ty<br>
&gt;&gt;&gt;&gt; considerations beyond those in 7315.<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; I accept that for the reduction part. But I'm not sure we =
can state <br>
&gt;&gt;&gt;&gt; that<br>
&gt;&gt;&gt;&gt; sort of thing for the expansion part, at least without som=
e more<br>
&gt;&gt;&gt;&gt; discussion. Since 7315 already acknowledges potential priv=
acy <br>
&gt;&gt;&gt;&gt; issues<br>
&gt;&gt;&gt;&gt; around P-Access-Network-Info, I'd like to at least see a s=
entence <br>
&gt;&gt;&gt;&gt; or two<br>
&gt;&gt;&gt;&gt; about the allowance of that in ACK requests, even if they =
just say <br>
&gt;&gt;&gt;&gt; that<br>
&gt;&gt;&gt;&gt; this addition makes things no worse than they already are.=
<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; OLD:<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; The security considerations for P- header fields are defined i=
n<br>
&gt;&gt;&gt;&nbsp;&nbsp; [RFC7315].&nbsp; This specification allows some he=
ader fields to be<br>
&gt;&gt;&gt;&nbsp;&nbsp; present in messages where they were previously not=
 allowed, and<br>
&gt;&gt;&gt;&nbsp;&nbsp; disallow some header fields to be present in messa=
ges where they <br>
&gt;&gt;&gt; were<br>
&gt;&gt;&gt;&nbsp;&nbsp; previously allowed. That does not cause any securi=
ty issues, but<br>
&gt;&gt;&gt;&nbsp;&nbsp; implementations need to be aware that implementati=
ons may not have<br>
&gt;&gt;&gt;&nbsp;&nbsp; been updated according to this document, and take =
proper actions <br>
&gt;&gt;&gt; if a<br>
&gt;&gt;&gt;&nbsp;&nbsp; header field occur, or does not occur, in a messag=
e where it <br>
&gt;&gt;&gt; should<br>
&gt;&gt;&gt;&nbsp;&nbsp; occur (or occurs in a message where it should not =
occur).<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; NEW:<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; The security considerations for these P- header fields are def=
ined <br>
&gt;&gt;&gt; in<br>
&gt;&gt;&gt;&nbsp;&nbsp; [RFC7315].&nbsp; This specification allows some he=
ader fields to be<br>
&gt;&gt;&gt;&nbsp;&nbsp; present in messages where they were previously not=
 allowed, and <br>
&gt;&gt;&gt; the<br>
&gt;&gt;&gt; security considerations and assumptions (e.g. regarding only s=
ending<br>
&gt;&gt;&gt; Information to trusted entities) also to those messages. In <b=
r>
&gt;&gt;&gt; addition,<br>
&gt;&gt;&gt; this specification also disallow some header fields to be pres=
ent<br>
&gt;&gt;&gt; in message where they were previously allowed. That does not c=
ause<br>
&gt;&gt;&gt; any security issues, but implementations need to be aware that=
<br>
&gt;&gt;&gt; implementations may not have been updated according to this <b=
r>
&gt;&gt;&gt; document,<br>
&gt;&gt;&gt; and take proper actions if a header field occur, or does not o=
ccur,<br>
&gt;&gt;&gt; in a message where it should occur (or occurs in a message whe=
re it<br>
&gt;&gt;&gt; should not occur).<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; Editorial:<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; -5, first sentence: &quot;The security considerations for =
P- header <br>
&gt;&gt;&gt;&gt; fields<br>
&gt;&gt;&gt;&gt; are defined in<br>
&gt;&gt;&gt;&gt;&nbsp;&nbsp; [RFC7315]&quot;<br>
&gt;&gt;&gt;&gt; I assume this means 7315 discusses the security considerat=
ions for <br>
&gt;&gt;&gt;&gt; these<br>
&gt;&gt;&gt;&gt; P-Headers specifically, not P-Headers in general. Is this =
the <br>
&gt;&gt;&gt;&gt; intent? If<br>
&gt;&gt;&gt;&gt; so, I suggest:<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; s/... for P-header fields.../ ... for these P-header field=
s...<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; I=B9ll fix as suggested (ass new text above).<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Regards,<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Christer<br>
</div>
</span></font>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B380FFF70ESESSMB209erics_--


From nobody Wed Jun 22 08:16:33 2016
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From: "Ben Campbell" <ben@nostrum.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>
Date: Wed, 22 Jun 2016 10:03:56 -0500
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Cc: SIPCORE <sipcore@ietf.org>, Gonzalo Salgueiro <gsalguei@cisco.com>, "draft-holmberg-dispatch-rfc7315-updates.all@ietf.org" <draft-holmberg-dispatch-rfc7315-updates.all@ietf.org>
Subject: Re: [sipcore] AD Evaluation of draft-holmberg-dispatch-rfc7315-updates-06
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Please go ahead. It might be worth replying to the IETF last call =

announcement with a mention of the update, in case anyone has a review =

in progress.

Thanks!

Ben.

On 22 Jun 2016, at 9:59, Christer Holmberg wrote:

> Good :)
>
> Is it ok of I submit a new version of the draft? That way the new text =

> will be available during  IETF last call.
>
> Regards,
>
> Christer
>
> Sent from my Windows Phone
> ________________________________
> From: Ben Campbell<mailto:ben@nostrum.com>
> Sent: =E2=80=8E22/=E2=80=8E06/=E2=80=8E2016 17:47
> To: Christer Holmberg<mailto:christer.holmberg@ericsson.com>
> Cc: Gonzalo Salgueiro<mailto:gsalguei@cisco.com>; =

> draft-holmberg-dispatch-rfc7315-updates.all@ietf.org<mailto:draft-holmb=
erg-dispatch-rfc7315-updates.all@ietf.org>; =

> SIPCORE<mailto:sipcore@ietf.org>
> Subject: Re: AD Evaluation of =

> draft-holmberg-dispatch-rfc7315-updates-06
>
> Works for me.
>
> Thanks!
>
> Ben.
>
> On 22 Jun 2016, at 2:18, Christer Holmberg wrote:
>
>> Hi,
>>
>> NEW:
>>
>>    =E2=80=9DThe security considerations for these P- header fields are=

>> defined in
>>    [RFC7315].  This specification allows some header fields to be
>>    present in messages where they were previously not allowed, and =

>> the
>>    security considerations and assumptions (e.g. regarding only
>> sending
>>    Information to trusted entities) also to those messages. In
>> addition,
>>    this specification also disallow some header fields to be present
>>    in message where they were previously allowed. That does not cause
>>    any security issues, but implementations need to be aware that
>>    implementations may not have been updated according to this
>> document,
>>    and take proper actions if a header field occur, or does not =

>> occur,
>>    in a message where it should occur (or occurs in a message where =

>> it
>>    should not occur). This document adds the ability to include
>>    P-Access-Network-Info in ACK requests. As documented in  =

>> [RFC7315],
>>    P-Access-Network-Info may include privacy sensitive information,
>> including
>>    the user's location. The security and privacy considerations for
>>    P-Access-Network-Info in ACK requests are similar to those for the
>> other
>>    SIP requests discussed in  [RFC7315].=E2=80=9D
>>
>> Regards,
>>
>> Christer
>>
>>
>> From: Christer Holmberg
>> <christer.holmberg@ericsson.com<mailto:christer.holmberg@ericsson.com>=
>
>> Date: Wednesday 22 June 2016 at 05:28
>> To: "gsalguei@cisco.com<mailto:gsalguei@cisco.com>"
>> <gsalguei@cisco.com<mailto:gsalguei@cisco.com>>, Ben Campbell
>> <ben@nostrum.com<mailto:ben@nostrum.com>>
>> Cc:
>> "draft-holmberg-dispatch-rfc7315-updates.all@ietf.org<mailto:draft-hol=
mberg-dispatch-rfc7315-updates.all@ietf.org>"
>> <draft-holmberg-dispatch-rfc7315-updates.all@ietf.org<mailto:draft-hol=
mberg-dispatch-rfc7315-updates.all@ietf.org>>,
>> "sipcore@ietf.org<mailto:sipcore@ietf.org>"
>> <sipcore@ietf.org<mailto:sipcore@ietf.org>>
>> Subject: RE: AD Evaluation of
>> draft-holmberg-dispatch-rfc7315-updates-06
>> Resent-From: <alias-bounces@ietf.org<mailto:alias-bounces@ietf.org>>
>> Resent-To: Christer Holmberg
>> <christer.holmberg@ericsson.com<mailto:christer.holmberg@ericsson.com>=
>,
>> Nevenka Biondic
>> <nevenka.biondic@ericsson.com<mailto:nevenka.biondic@ericsson.com>>,
>> "gsalguei@cisco.com<mailto:gsalguei@cisco.com>"
>> <gsalguei@cisco.com<mailto:gsalguei@cisco.com>>, Ben Campbell
>> <ben@nostrum.com<mailto:ben@nostrum.com>>, "A. Mahoney"
>> <mahoney@nostrum.com<mailto:mahoney@nostrum.com>>
>> Resent-Date: Wednesday 22 June 2016 at 05:28
>>
>> Hi,
>>
>> We can add the text.
>>
>> Regards,
>>
>> Christer
>>
>> Sent from my Windows Phone
>> ________________________________
>> From: Gonzalo Salgueiro (gsalguei)<mailto:gsalguei@cisco.com>
>> Sent: 21/06/2016 19:44
>> To: Ben Campbell<mailto:ben@nostrum.com>
>> Cc: Christer Holmberg<mailto:christer.holmberg@ericsson.com>;
>> draft-holmberg-dispatch-rfc7315-updates.all@ietf.org<mailto:draft-holm=
berg-dispatch-rfc7315-updates.all@ietf.org>;
>> SIPCORE<mailto:sipcore@ietf.org>
>> Subject: Re: AD Evaluation of
>> draft-holmberg-dispatch-rfc7315-updates-06
>>
>>
>>> On Jun 21, 2016, at 11:22 AM, Ben Campbell
>>> <ben@nostrum.com<mailto:ben@nostrum.com>> wrote:
>>>
>>> That's a good start, but don't be surprised if we get questions
>>> specifically about adding NPLI to ACK requests. some language to the
>>> effect of the following might help:
>>>
>>> "This document adds the ability to include P-Access-Network-Info in
>>> ACK requests. As documented in RFC7315, P-Access-Network-Info may
>>> include privacy sensitive information, including the user's =

>>> location.
>>> The security and privacy considerations for P-Access-Network-Info in
>>> ACK requests are similar to those for the other SIP requests
>>> discussed in RFC7315.=E2=80=9D
>>
>> I=E2=80=99m fine with adding such text.
>>
>> Christer - Can we append this to your proposed text?
>>
>> Gonzalo
>>
>>
>>>
>>> Thanks!
>>>
>>> Ben.
>>>
>>> On 21 Jun 2016, at 3:26, Christer Holmberg wrote:
>>>
>>>> Hi Ben,
>>>>
>>>> See inline.
>>>>
>>>>> --------------
>>>>>
>>>>> Substantive:
>>>>>
>>>>> The security considerations state that the draft removes some
>>>>> places
>>>>> that some of the P-Headers can be sent, but expands that to some
>>>>> other
>>>>> places. Further, it says that neither introduce new security
>>>>> considerations beyond those in 7315.
>>>>>
>>>>> I accept that for the reduction part. But I'm not sure we can =

>>>>> state
>>>>> that
>>>>> sort of thing for the expansion part, at least without some more
>>>>> discussion. Since 7315 already acknowledges potential privacy
>>>>> issues
>>>>> around P-Access-Network-Info, I'd like to at least see a sentence
>>>>> or two
>>>>> about the allowance of that in ACK requests, even if they just say
>>>>> that
>>>>> this addition makes things no worse than they already are.
>>>>
>>>>
>>>> OLD:
>>>>
>>>> The security considerations for P- header fields are defined in
>>>>   [RFC7315].  This specification allows some header fields to be
>>>>   present in messages where they were previously not allowed, and
>>>>   disallow some header fields to be present in messages where they
>>>> were
>>>>   previously allowed. That does not cause any security issues, but
>>>>   implementations need to be aware that implementations may not =

>>>> have
>>>>   been updated according to this document, and take proper actions
>>>> if a
>>>>   header field occur, or does not occur, in a message where it
>>>> should
>>>>   occur (or occurs in a message where it should not occur).
>>>>
>>>>
>>>>
>>>> NEW:
>>>>
>>>> The security considerations for these P- header fields are defined
>>>> in
>>>>   [RFC7315].  This specification allows some header fields to be
>>>>   present in messages where they were previously not allowed, and
>>>> the
>>>> security considerations and assumptions (e.g. regarding only =

>>>> sending
>>>> Information to trusted entities) also to those messages. In
>>>> addition,
>>>> this specification also disallow some header fields to be present
>>>> in message where they were previously allowed. That does not cause
>>>> any security issues, but implementations need to be aware that
>>>> implementations may not have been updated according to this
>>>> document,
>>>> and take proper actions if a header field occur, or does not occur,
>>>> in a message where it should occur (or occurs in a message where it
>>>> should not occur).
>>>>
>>>>
>>>>
>>>>> Editorial:
>>>>>
>>>>> -5, first sentence: "The security considerations for P- header
>>>>> fields
>>>>> are defined in
>>>>>   [RFC7315]"
>>>>> I assume this means 7315 discusses the security considerations for
>>>>> these
>>>>> P-Headers specifically, not P-Headers in general. Is this the
>>>>> intent? If
>>>>> so, I suggest:
>>>>>
>>>>> s/... for P-header fields.../ ... for these P-header fields...
>>>>
>>>> I=C2=B9ll fix as suggested (ass new text above).
>>>>
>>>> Regards,
>>>>
>>>> Christer


From nobody Wed Jun 22 09:58:30 2016
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Ben Campbell <ben@nostrum.com>
Thread-Topic: AD Evaluation of draft-holmberg-dispatch-rfc7315-updates-06
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Cc: SIPCORE <sipcore@ietf.org>, Gonzalo Salgueiro <gsalguei@cisco.com>, "draft-holmberg-dispatch-rfc7315-updates.all@ietf.org" <draft-holmberg-dispatch-rfc7315-updates.all@ietf.org>
Subject: Re: [sipcore] AD Evaluation of draft-holmberg-dispatch-rfc7315-updates-06
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From: Roman Shpount <roman@telurix.com>
Date: Wed, 22 Jun 2016 17:02:41 -0400
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Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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Rifaat,

In your example, I would expect to issue an HTTP request to create the
conference, get the SIP URI back (with authentication token embedded in the
URI) and place the SIP call to this URI. I do not see a reason why
any standardization effort is required to implement this.

Regards,

_____________
Roman Shpount

On Wed, Jun 22, 2016 at 8:48 AM, Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
wrote:

> Roman,
>
> I think that the question of how to exactly carry the token in SIP request
> is a minor issue *at this stage*.
> We are discussing the idea and concept of delegating both the AuthN and
> AuthZ to separate server(s) and how to achieve that.
>
> Regards,
>  Rifaat
>
>
> On Tue, Jun 21, 2016 at 7:50 PM, Roman Shpount <roman@telurix.com> wrote:
>
>> Furthermore, reading all of this, I cannot understand why Authorization
>> token cannot be a part of SIP URL, either as user part or URL parameter.
>> This does not need to be anything more then service specific URL and does
>> not require a new SIP header or authentication mechanism.
>>
>> _____________
>> Roman Shpount
>>
>> On Tue, Jun 21, 2016 at 7:46 PM, Peterson, Jon <jon.peterson@neustar.biz>
>> wrote:
>>
>>>
>>> It's not really what I'm looking for, no, and after mulling this over a
>>> bit, I fear going farther along this path would only drill down into levels
>>> of complexity that will likely make this harder to understand rather than
>>> easier.
>>>
>>> I do acknowledge that you describe the INVITE as being rejected if it
>>> doesn't contain a token - but that's not the same thing as saying the
>>> INVITE will be rejected because the relying party makes an authorization
>>> decision based on the attributes in the token. It's more like an INVITE
>>> being rejected because a user agent can't supply an Authorization header. I
>>> will reiterate that I don't think the kinds of attributes you are proposing
>>> should be in the scope of an effort that will try to address a framework
>>> for authorizing SIP requests. I don't think the values of the attributes
>>> (as opposed to the mere presence of some attributes) have anything to do
>>> with authorizing the SIP request that you propose would carry the
>>> attributes.
>>>
>>> With regards to this whole conferencing example, it does seem a bit
>>> contrived to me, like you're sketching a SIP-based way of setting
>>> conference policy on the fly that happens to require the kind of attributes
>>> you want, and then saying that demonstrates the need for the attributes.
>>> The fact that all of the ways to do conference policy we can point to in
>>> fact do it via the web makes it hard for me to understand why we should
>>> create a SIP-based way of doing this. To me, it looks like you're opening a
>>> problem space that has much more to do with reshaping the way that SIP does
>>> conferences (and again, as Radhika's mail pointing out, there's been a lot
>>> of thinking about this since the days of RFC4579) than it does with
>>> defining a framework for authorizing SIP requests.
>>>
>>> Jon Peterson
>>> Neustar, Inc.
>>>
>>> From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
>>> Date: Thursday, June 16, 2016 at 6:19 AM
>>> To: Jon Peterson <jon.peterson@neustar.biz>
>>> Cc: "sipcore@ietf.org" <sipcore@ietf.org>
>>> Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
>>>
>>> The attributes I provided are examples of attributes that might be
>>> provided in an access token; it was not meant to include all the attributes
>>> for this use case, because my intention was not to focus on conference
>>> server, but to discuss the idea in general.
>>>
>>> The following could be used with the conference factory to request a
>>> conference focus:
>>> The INVITE sent to the conference factory could include a token that has
>>> a * conference-type* attribute which could take one of the following
>>> values: *audio, video, webconference, audio-webconference, or
>>> video-webconference*.
>>> The conference factory will then issue a *conference uri* for the
>>> requested conference type, assuming the token is valid. If the
>>> *conference-type* attribute is missing, then the request would be
>>> rejected.
>>>
>>> Is that what you are looking for?
>>>
>>> Regards,
>>>  Rifaat
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Jun 15, 2016 at 5:06 PM, Peterson, Jon <jon.peterson@neustar.biz
>>> > wrote:
>>>
>>>>
>>>>>
>>>>> In that case, if you could furnish a use case with attributes that
>>>>> would be carried in a SIP iNVITE that are actually salient to the
>>>>> authorization of that particular SIP request, we'd be getting a lot closer
>>>>> to common ground.
>>>>>
>>>>>
>>>> An obvious example would be an INVITE with a token sent to a conference
>>>> factory to authorize the creation of a focus.
>>>>
>>>>
>>>> We're still talking past each other. My point above was that the two
>>>> attributes you specified - constraints on conference size and media types -
>>>> have nothing to do with the authorization of the INVITE carrying those
>>>> attributes. Like, unless the conference size permitted is 0, then the
>>>> INVITE that creates an ad hoc conference and connects you to its bridge is
>>>> not going to be rejected by the focus on the basis of that conference size
>>>> attribute. If the maximum conference size were 0, the AuthZ service just
>>>> wouldn't attach a token to an INVITE as it would never be allowed to start
>>>> a conference. Same goes for media types.
>>>>
>>>> Instead, what your two attributes constrain is what happens after the
>>>> conference is created: when ten users try to dial in to a conference with a
>>>> permitted size of nine, say. That tenth INVITE would get rejected - not the
>>>> conference-forming INVITE that you propose would carry the token. That's
>>>> why what you're proposing is overloading the INVITE to tunnel conference
>>>> policy constraints to the focus. I mean, maybe that's a problem space worth
>>>> investigating, though I suspect there are good reasons why RFC4353
>>>> specified that conference policy is not manipulated with SIP - mostly
>>>> because the semantics of manipulating conference policy probably don't map
>>>> well onto dialog-forming SIP requests. But I think that conference policy
>>>> provisioning problem space would be pretty far from where we started in
>>>> this discussion.
>>>>
>>>> Let me reiterate what I wrote in my previous mail, which I quote here
>>>> at the top: "furnish a use case with attributes that would be carried in a
>>>> SIP iNVITE that are actually salient to the authorization of that
>>>> particular SIP request." Again: "that particular SIP request." Not later
>>>> transactions in some broader context in which the original user agent
>>>> probably isn't even an endpoint.
>>>>
>>>> The Authorization header in SIP, for example, is actually used by the
>>>> UAS to make a decision about whether or not to authorize the request in
>>>> which the header appears.  The token that you propose a SIP INVITE would
>>>> carry in this case is not used by the relying party to make a decision
>>>> about whether or not that INVITE is authorized. That's why I believe this
>>>> conference policy issue is a different problem space entirely from figuring
>>>> out how to implement attribute-based authorization for SIP requests.
>>>>
>>>> Jon Peterson
>>>> Neustar, Inc.
>>>>
>>>>
>>>> Regards,
>>>>  Rifaat
>>>>
>>>>
>>>
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>
>>>
>>
>

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<div dir=3D"ltr"><span style=3D"color:rgb(0,0,0);font-size:12.8px">Rifaat,<=
/span><br><div><span style=3D"color:rgb(0,0,0);font-size:12.8px"><br></span=
></div><div><font color=3D"#000000"><span style=3D"font-size:12.8px">In you=
r example, I would expect to issue an HTTP request to create the conference=
, get the SIP URI back (with authentication token embedded in the URI) and =
place the SIP call to this URI. I do not see a reason why any=C2=A0standard=
ization=C2=A0effort is required to implement this.</span></font></div><div>=
<font color=3D"#000000"><span style=3D"font-size:12.8px"><br></span></font>=
</div><div><font color=3D"#000000"><span style=3D"font-size:12.8px">Regards=
,</span></font></div></div><div class=3D"gmail_extra"><br clear=3D"all"><di=
v><div class=3D"gmail_signature" data-smartmail=3D"gmail_signature">_______=
______<br>Roman Shpount</div></div>
<br><div class=3D"gmail_quote">On Wed, Jun 22, 2016 at 8:48 AM, Rifaat Shek=
h-Yusef <span dir=3D"ltr">&lt;<a href=3D"mailto:rifaat.ietf@gmail.com" targ=
et=3D"_blank">rifaat.ietf@gmail.com</a>&gt;</span> wrote:<br><blockquote cl=
ass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;p=
adding-left:1ex"><div dir=3D"ltr">Roman,<div><br></div><div>I think that th=
e question of how to exactly carry the token in SIP request is a minor issu=
e <b>at this stage</b>.<br></div><div>We are discussing the idea and concep=
t of delegating both the AuthN and AuthZ to separate server(s) and how to a=
chieve that.<br></div><div><br></div><div>Regards,</div><div>=C2=A0Rifaat</=
div><div><br></div></div><div class=3D"HOEnZb"><div class=3D"h5"><div class=
=3D"gmail_extra"><br><div class=3D"gmail_quote">On Tue, Jun 21, 2016 at 7:5=
0 PM, Roman Shpount <span dir=3D"ltr">&lt;<a href=3D"mailto:roman@telurix.c=
om" target=3D"_blank">roman@telurix.com</a>&gt;</span> wrote:<br><blockquot=
e class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc sol=
id;padding-left:1ex"><div dir=3D"ltr">Furthermore, reading all of this, I c=
annot understand why Authorization token cannot be a part of SIP URL, eithe=
r as user part or URL parameter. This does not need to be anything more the=
n service specific URL and does not require a new SIP header or authenticat=
ion mechanism.</div><div class=3D"gmail_extra"><br clear=3D"all"><div><div =
data-smartmail=3D"gmail_signature">_____________<br>Roman Shpount</div></di=
v>
<br><div class=3D"gmail_quote"><div><div>On Tue, Jun 21, 2016 at 7:46 PM, P=
eterson, Jon <span dir=3D"ltr">&lt;<a href=3D"mailto:jon.peterson@neustar.b=
iz" target=3D"_blank">jon.peterson@neustar.biz</a>&gt;</span> wrote:<br></d=
iv></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;borde=
r-left:1px #ccc solid;padding-left:1ex"><div><div>



<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>It&#39;s not really what I&#39;m looking for, no, and after mulling th=
is over a bit, I fear going farther along this path would only drill down i=
nto levels of complexity that will likely make this harder to understand ra=
ther than easier.</div>
<div><br>
</div>
<div>I do acknowledge that you describe the INVITE as being rejected if it =
doesn&#39;t contain a token - but that&#39;s not the same thing as saying t=
he INVITE will be rejected because the relying party makes an authorization=
 decision based on the attributes in the
 token. It&#39;s more like an INVITE being rejected because a user agent ca=
n&#39;t supply an Authorization header. I will reiterate that I don&#39;t t=
hink the kinds of attributes you are proposing should be in the scope of an=
 effort that will try to address a framework
 for authorizing SIP requests. I don&#39;t think the values of the attribut=
es (as opposed to the mere presence of some attributes) have anything to do=
 with authorizing the SIP request that you propose would carry the attribut=
es.</div>
<div><br>
</div>
<div>
<div>With regards to this whole conferencing example, it does seem a bit co=
ntrived to me, like you&#39;re sketching a SIP-based way of setting confere=
nce policy on the fly that happens to require the kind of attributes you wa=
nt, and then saying that demonstrates
 the need for the attributes. The fact that all of the ways to do conferenc=
e policy we can point to in fact do it via the web makes it hard for me to =
understand why we should create a SIP-based way of doing this. To me, it lo=
oks like you&#39;re opening a problem
 space that has much more to do with reshaping the way that SIP does confer=
ences (and again, as Radhika&#39;s mail pointing out, there&#39;s been a lo=
t of thinking about this since the days of RFC4579) than it does with defin=
ing a framework for authorizing SIP requests.</div>
</div>
<div><br>
</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-BOTTOM:medium none;BORDER-LEFT:medium none;PADDING-BOTTOM:0in;PADD=
ING-LEFT:0in;PADDING-RIGHT:0in;BORDER-TOP:#b5c4df 1pt solid;BORDER-RIGHT:me=
dium 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" target=3D"_blank">rifaat.ietf@gmail.com<=
/a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Thursday, June 16, 2016 at 6:=
19 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>&quot;<a href=3D"mailto:sipcore=
@ietf.org" target=3D"_blank">sipcore@ietf.org</a>&quot; &lt;<a href=3D"mail=
to:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [sipcore] SIP AuthNZ P=
roblem Statement - v3<br>
</div>
<div><br>
</div>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div>
<div>
<div dir=3D"ltr">
<div>The attributes I provided are examples of attributes that might be pro=
vided in an access token; it was not meant to include all the attributes fo=
r this use case, because my intention was not to focus on conference server=
, but to discuss the idea in general.</div>
<div><br>
</div>
<div>
<div>The following could be used with the conference factory to request a c=
onference focus:</div>
<div>The INVITE sent to the conference factory could include a token that h=
as a <b>
conference-type</b> attribute which could take one of the following values:=
 <b>audio, video, webconference, audio-webconference, or video-webconferenc=
e</b>.</div>
<div>The conference factory will then issue a <b>conference uri</b> for the=
 requested conference type, assuming the token is valid. If the
<b>conference-type</b> attribute is missing, then the request would be reje=
cted.</div>
</div>
<div><br>
</div>
<div>Is that what you are looking for?</div>
<div><br>
</div>
<div>Regards,</div>
<div>=C2=A0Rifaat</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Wed, Jun 15, 2016 at 5:06 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:0 0 0 .8ex;border-left:1p=
x #ccc 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><span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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">
<span><br>
<br>
</span>
<div>
<div>In that case, if you could furnish a use case with attributes that wou=
ld be carried in a SIP iNVITE that are actually salient to the authorizatio=
n of that particular SIP request, we&#39;d be getting a lot closer to commo=
n ground.</div>
<div><br>
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>An obvious example would be an INVITE with a token sent to a conferenc=
e factory to authorize the creation of a focus.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span>
<div>We&#39;re still talking past each other. My point above was that the t=
wo attributes you specified - constraints on conference size and media type=
s - have nothing to do with the authorization of the INVITE carrying those =
attributes. Like, unless the conference
 size permitted is 0, then the INVITE that creates an ad hoc conference and=
 connects you to its bridge is not going to be rejected by the focus on the=
 basis of that conference size attribute. If the maximum conference size we=
re 0, the AuthZ service just wouldn&#39;t
 attach a token to an INVITE as it would never be allowed to start a confer=
ence. Same goes for media types.</div>
<div><br>
</div>
<div>Instead, what your two attributes constrain is what happens after the =
conference is created: when ten users try to dial in to a conference with a=
 permitted size of nine, say. That tenth INVITE would get rejected - not th=
e conference-forming INVITE that
 you propose would carry the token. That&#39;s why what you&#39;re proposin=
g is overloading the INVITE to tunnel conference policy constraints to the =
focus. I mean, maybe that&#39;s a problem space worth investigating, though=
 I suspect there are good reasons why RFC4353
 specified that conference policy is not manipulated with SIP - mostly beca=
use the semantics of manipulating conference policy probably don&#39;t map =
well onto dialog-forming SIP requests. But I think that conference policy p=
rovisioning problem space would be pretty
 far from where we started in this discussion.</div>
<div><br>
</div>
<div>Let me reiterate what I wrote in my previous mail, which I quote here =
at the top: &quot;furnish a use case with attributes that would be carried =
in a SIP iNVITE that are actually salient to the authorization of that part=
icular SIP request.&quot; Again: &quot;that particular
 SIP request.&quot; Not later transactions in some broader context in which=
 the original user agent probably isn&#39;t even an endpoint.</div>
<div><br>
</div>
<div>The Authorization header in SIP, for example, is actually used by the =
UAS to make a decision about whether or not to authorize the request in whi=
ch the header appears.=C2=A0 The token that you propose a SIP INVITE would =
carry in this case is not used by the
 relying party to make a decision about whether or not that INVITE is autho=
rized. That&#39;s why I believe this conference policy issue is a different=
 problem space entirely from figuring out how to implement attribute-based =
authorization for SIP requests.</div>
<span>
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<div>Regards,</div>
<div>=C2=A0Rifaat</div>
<div><br>
</div>
</div>
</div>
</div>
</blockquote>
</span></span></div>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</blockquote>
</span>
</div>

<br></div></div><span>_______________________________________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a><=
br>
</span><a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" rel=3D"nor=
eferrer" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sipcore</a=
><br>
<br></blockquote></div><br></div>
</blockquote></div><br></div>
</div></div></blockquote></div><br></div>

--001a113ed0d02f8e160535e44219--


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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Thread-Topic: [sipcore] SIP AuthNZ Problem Statement - v3
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Date: Wed, 22 Jun 2016 22:29:18 +0000
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The token added to the INVITE is not a policy; it is an authorization by th=
e AuthZ Server to allow this request to create a conference focus.

Well, this is the complexity level I was trying to avoid, but, I do agree t=
hat what your proposed token carries isn't strictly a conference policy - i=
t is instead a policy that constrains conference policies that users are al=
lowed to set.

The thing that RFC4353 calls "conference policy" is essentially a policy fo=
r a particular instance of a conference. The two attributes you initially p=
roposed actually constrain conference policies rather than conferences - th=
at is, they apply to all conferences that I might try to set policy for, ra=
ther than any given conference instance. When I set up a particular confere=
nce, for example, I might want it to be voice only, and to have four seats.=
 That's the policy I want for that conference. I gather the attributes you =
proposed are the constraints that my local administrative domain imposes on=
 conferences I create, so the conference policy I set for any given policy =
needs to be a subset of those constraints. If my administrative domain limi=
ts me to only three-seat conferences, then when I try to set a conference p=
olicy for four seats, that manipulation of conference policy would presumab=
ly be rejected when it is compared to my administrative domain's constraint=
s.

But of course, in the ad hoc corner case you are considering, the conferenc=
e bridge has no way of anticipating how many users Alice will try to add to=
 the conference - so, her initial INVITE containing the parameter would nev=
er be rejected for violating the constraint on conference size, only later =
INVITEs would be. That attribute has nothing to do with authorizing the INV=
ITE carrying the token.

Moreover, how does one actually manipulate my conference policy? According =
to RFC4353, it isn't with SIP. Probably with the web, with an HTTP request.=
 RFC4579 et al do not provide provisions for setting conference policy. Add=
ing the attributes you proposed to SIP isn't really about setting conferenc=
e policy, it's about alerting the conference service to the constraints on =
conference policy that are imposed by my administrative domain. This is an =
important distinction, because it shows how far removed that SIP transactio=
n actually is from the way conference policy is set. Why would I send a SIP=
 request with the constraints on conference policy if I'm only going to hav=
e to use HTTP to actually set conference policy later? Why wouldn't my admi=
nistrative domain have me attach its imposed constraints on conference poli=
cy to the actual HTTP message that is manipulating conference policy?

<snip>

Now, Alice decides to start an audio conference; to do that, the phone obta=
ins an access token from the AuthNZ server that allows Alice to start a con=
ference of the type =93audio=94.

I did figure you had something like this is mind - perhaps that Alice would=
 send her INVITE to the IdP first, and that the IdP would then  inspect and=
 authorize a conference policy that Alice somehow conveyed via the INVITE. =
Through what field or parameter do you suppose to INVITE will convey that A=
lice intends for the conference to be restricted to audio media? Or do you =
perhaps mean that Alice would use HTTP to inform her IdP of the media choic=
es for the conference - in the which case policy for the conference starts =
in the HTTP realm, and should certainly stay there.

I imagine that in the corner case of an ad hoc conference, the INVITE that =
Alice is sending would contain some session negotiation parameters that cou=
ld be taken to imply what sorts of media types Alice wants to allow or forb=
id in the conference. But I will reiterate that RFC4579 and the related spe=
cifications I can think of off the top of my head do not contain a provisio=
n for the INVITE to communicate session policy information - and I don't th=
ink it is safe to infer from the SDP in an INVITE that Alice intends to lim=
it the conference to the media type(s) it offers.

Moreover, in the ad hoc conference corner case, I see no plausible justific=
ation for why the conference bridge would ever reject Alice's initial INVIT=
E based on the media type it contains. If she's sending her INVITE through =
the IdP via SIP, it would never even forward the request, let alone issue a=
 token for it, if it didn't like the media type that Alice was proposing - =
if Alice proposes "video", and the IdP only allows "audio", when the IdP re=
ceives Alice's INVITE containing "video" it would do the rejecting itself (=
in SIP or HTTP). So once again, I don't see how this token would ever lead =
to the actual INVITE that carried the token ever being rejected by the conf=
erence bridge based on the media types authorized by the token.

Jon Peterson
Neustar, Inc.

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252">
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<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>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div>
<div>
<div dir=3D"ltr">
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%"><font face=3D"monospace,monospace">The token a=
dded to the INVITE is
<b>not a policy</b>; it is an authorization by the AuthZ Server to allow th=
is request to create a conference focus.</font></span></p>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>Well, this is the complexity level I was trying to avoid, but, I do ag=
ree that what your proposed token carries isn't strictly a conference polic=
y - it is instead a policy that constrains conference policies that users a=
re allowed to set.</div>
<div><br>
</div>
<div>The thing that RFC4353 calls &quot;conference policy&quot; is essentia=
lly a policy for a particular instance of a conference. The two attributes =
you initially proposed actually constrain conference policies rather than c=
onferences - that is, they apply to all conferences
 that I might try to set policy for, rather than any given conference insta=
nce. When I set up a particular conference, for example, I might want it to=
 be voice only, and to have four seats. That's the policy I want for that c=
onference. I gather the attributes
 you proposed are the constraints that my local administrative domain impos=
es on conferences I create, so the conference policy I set for any given po=
licy needs to be a subset of those constraints. If my administrative domain=
 limits me to only three-seat conferences,
 then when I try to set a conference policy for four seats, that manipulati=
on of conference policy would presumably be rejected when it is compared to=
 my administrative domain's constraints.&nbsp;</div>
<div><br>
</div>
<div>But of course, in the ad hoc corner case you are considering, the conf=
erence bridge has no way of anticipating how many users Alice will try to a=
dd to the conference - so, her initial INVITE containing the parameter woul=
d never be rejected for violating
 the constraint on conference size, only later INVITEs would be. That attri=
bute has nothing to do with authorizing the INVITE carrying the token.</div=
>
<div><br>
</div>
<div>Moreover, how does one actually manipulate my conference policy? Accor=
ding to RFC4353, it isn't with SIP. Probably with the web, with an HTTP req=
uest. RFC4579 et al do not provide provisions for setting conference policy=
. Adding the attributes you proposed
 to SIP isn't really about setting conference policy, it's about alerting t=
he conference service to the constraints on conference policy that are impo=
sed by my administrative domain. This is an important distinction, because =
it shows how far removed that SIP
 transaction actually is from the way conference policy is set. Why would I=
 send a SIP request with the constraints on conference policy if I'm only g=
oing to have to use HTTP to actually set conference policy later? Why would=
n't my administrative domain have
 me attach its imposed constraints on conference policy to the actual HTTP =
message that is manipulating conference policy?</div>
<div><br>
</div>
<div>&lt;snip&gt;</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<p class=3D"MsoNormal" style=3D"margin-bottom: 0.0001pt;"><span style=3D"fo=
nt-size: 10pt; line-height: 15.333332061767578px; background-repeat: initia=
l initial;"><font face=3D"monospace,monospace"><br>
Now, Alice decides to start an audio conference; to do that, the phone obta=
ins an access token from the AuthNZ server that allows Alice to start a con=
ference of the type =93<b>audio=94</b>.</font></span></p>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>I did figure you had something like this is mind - perhaps that Alice =
would send her INVITE to the IdP first, and that the IdP would then &nbsp;i=
nspect and authorize a conference policy that Alice somehow conveyed via th=
e INVITE. Through what field or parameter
 do you suppose to INVITE will convey that Alice intends for the conference=
 to be restricted to audio media? Or do you perhaps mean that Alice would u=
se HTTP to inform her IdP of the media choices for the conference - in the =
which case policy for the conference
 starts in the HTTP realm, and should certainly stay there.</div>
<div><br>
</div>
<div>I imagine that in the corner case of an ad hoc conference, the INVITE =
that Alice is sending would contain some session negotiation parameters tha=
t could be taken to imply what sorts of media types Alice wants to allow or=
 forbid in the conference. But I
 will reiterate that RFC4579 and the related specifications I can think of =
off the top of my head do not contain a provision for the INVITE to communi=
cate session policy information - and I don't think it is safe to infer fro=
m the SDP in an INVITE that Alice
 intends to limit the conference to the media type(s) it offers.</div>
<div><br>
</div>
<div>Moreover, in the ad hoc conference corner case, I see no plausible jus=
tification for why the conference bridge would ever reject Alice's initial =
INVITE based on the media type it contains. If she's sending her INVITE thr=
ough the IdP via SIP, it would never
 even forward the request, let alone issue a token for it, if it didn't lik=
e the media type that Alice was proposing - if Alice proposes &quot;video&q=
uot;, and the IdP only allows &quot;audio&quot;, when the IdP receives Alic=
e's INVITE containing &quot;video&quot; it would do the rejecting
 itself (in SIP or HTTP). So once again, I don't see how this token would e=
ver lead to the actual INVITE that carried the token ever being rejected by=
 the conference bridge based on the media types authorized by the token.</d=
iv>
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
</body>
</html>

--_000_D390592919FF98jonpetersonneustarbiz_--


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From: "Roy, Radhika R CIV USARMY RDECOM CERDEC (US)" <radhika.r.roy.civ@mail.mil>
To: "Peterson, Jon" <jon.peterson@neustar.biz>, Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Thread-Topic: [Non-DoD Source] Re: [sipcore] SIP AuthNZ Problem Statement - v3
Thread-Index: AQHRqS0TDyFOI6cZKUWBz4pHTe07DJ+vbNaAgAFjyQCAA2UIgIACvqkAgANXvoCABITogIAH9hmAgAMlg4CAF/VngIAF7xoAgAGiQYD//8wQAIAB3CEA///zYgCAAYUpAIAIFXSAgAFN74CAAC7vAIABEPqg
Date: Thu, 23 Jun 2016 11:55:36 +0000
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References: <CAGL6ep+8nhiuQs8uBLxmsdwfrxFys4PM8UP5Apbcb7YVK4fmfg@mail.gmail.com> <CAGL6epKhbrdNEgpLFDD+JHFCNcAfST1aCKg3votcVe6z4wVa6A@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B37F97548@ESESSMB209.ericsson.se> <D3562523.18AAA1%jon.peterson@neustar.biz> <CAGL6epJDu1ojsSaPnJqpNoZgCwXvOoFNoEa2ZLNtmRtKMPMyoA@mail.gmail.com> <D35B4CE3.18D9AB%jon.peterson@neustar.biz> <CAGL6epJuByqHi_Fvssyd0V-LE4=_s7HK9DB-9Z3SSVky1Ak4Xw@mail.gmail.com> <D361F10A.18F366%jon.peterson@neustar.biz> <CAGL6epKVaOyVMS_6xmdEXbWsp2UXsbbwjTW2MgM=fJWLE381dg@mail.gmail.com> <D3688271.191B87%jon.peterson@neustar.biz> <CAGL6epKiY_Xi_D+J8QnTfoEyaDU1=jfcbhVFAMY_v3tkQjAhug@mail.gmail.com> <D3842CA2.19BA97%jon.peterson@neustar.biz> <CAGL6epKbdjHP-K9Hjn825HUVTQ4V=TnUYeyrLSU0ETQtmsfs9g@mail.gmail.com> <D385862C.19C10C%jon.peterson@neustar.biz> <CAGL6epLZOhUMQHGGW_Y49gZZAdvFWH07NfO97f=krk1_K9Xd-g@mail.gmail.com> <D386CD3B.19C95F%jon.peterson@neustar.biz> <CAGL6epKj00is8dH7NHdjjuTbh-XbeB7yuALouzR2fzfMBC=R4w@mail.gmail.com> <D38F19C3.19F959%jon.peterson@neustar.biz> <CAGL6epK_DEPr5nYOfGOszLtfT8i8Lso8cZkdXO2+v613ix3eDQ@mail.gmail.com> <D3905929.19FF98%jon.peterson@neustar.biz>
In-Reply-To: <D3905929.19FF98%jon.peterson@neustar.biz>
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Subject: Re: [sipcore] [Non-DoD Source] Re: SIP AuthNZ Problem Statement - v3
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Hi, Jon:

"So once again, I don't see how this token would ever lead to the actual IN=
VITE that carried the token ever being rejected by the conference bridge ba=
sed on the media types authorized by the token."

Based on your above conclusion, it appears that there is something missing =
the way the SIP conferencing is set up. My guess is that both the conferenc=
e initiator (who sent the original INVITE) and conference "focus" should be=
 autonomous in accepting or rejecting the call if the conferencing paramete=
rs are not within the expectation of their respective domains.

I would be glad to know what are the criteria that the SIP-based conferenci=
ng based on focus standardized so far for accepting or rejecting the call i=
f it is not token or it is other parameter including SDP.

Your analysis will really be helpful to us.

BR/Radhika

-----Original Message-----
From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Peterson, Jon
Sent: Wednesday, June 22, 2016 6:29 PM
To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Cc: sipcore@ietf.org
Subject: [Non-DoD Source] Re: [sipcore] SIP AuthNZ Problem Statement - v3


	The token added to the INVITE is not a policy; it is an authorization by t=
he AuthZ Server to allow this request to create a conference focus.


Well, this is the complexity level I was trying to avoid, but, I do agree t=
hat what your proposed token carries isn't strictly a conference policy - i=
t is instead a policy that constrains conference policies that users are al=
lowed to set.

The thing that RFC4353 calls "conference policy" is essentially a policy fo=
r a particular instance of a conference. The two attributes you initially p=
roposed actually constrain conference policies rather than conferences - th=
at is, they apply to all conferences that I might try to set policy for, ra=
ther than any given conference instance. When I set up a particular confere=
nce, for example, I might want it to be voice only, and to have four seats.=
 That's the policy I want for that conference. I gather the attributes you =
proposed are the constraints that my local administrative domain imposes on=
 conferences I create, so the conference policy I set for any given policy =
needs to be a subset of those constraints. If my administrative domain limi=
ts me to only three-seat conferences, then when I try to set a conference p=
olicy for four seats, that manipulation of conference policy would presumab=
ly be rejected when it is compared to my administrative domain's constraint=
s.=20

But of course, in the ad hoc corner case you are considering, the conferenc=
e bridge has no way of anticipating how many users Alice will try to add to=
 the conference - so, her initial INVITE containing the parameter would nev=
er be rejected for violating the constraint on conference size, only later =
INVITEs would be. That attribute has nothing to do with authorizing the INV=
ITE carrying the token.

Moreover, how does one actually manipulate my conference policy? According =
to RFC4353, it isn't with SIP. Probably with the web, with an HTTP request.=
 RFC4579 et al do not provide provisions for setting conference policy. Add=
ing the attributes you proposed to SIP isn't really about setting conferenc=
e policy, it's about alerting the conference service to the constraints on =
conference policy that are imposed by my administrative domain. This is an =
important distinction, because it shows how far removed that SIP transactio=
n actually is from the way conference policy is set. Why would I send a SIP=
 request with the constraints on conference policy if I'm only going to hav=
e to use HTTP to actually set conference policy later? Why wouldn't my admi=
nistrative domain have me attach its imposed constraints on conference poli=
cy to the actual HTTP message that is manipulating conference policy?

<snip>

=09
	Now, Alice decides to start an audio conference; to do that, the phone obt=
ains an access token from the AuthNZ server that allows Alice to start a co=
nference of the type "audio".


I did figure you had something like this is mind - perhaps that Alice would=
 send her INVITE to the IdP first, and that the IdP would then  inspect and=
 authorize a conference policy that Alice somehow conveyed via the INVITE. =
Through what field or parameter do you suppose to INVITE will convey that A=
lice intends for the conference to be restricted to audio media? Or do you =
perhaps mean that Alice would use HTTP to inform her IdP of the media choic=
es for the conference - in the which case policy for the conference starts =
in the HTTP realm, and should certainly stay there.

I imagine that in the corner case of an ad hoc conference, the INVITE that =
Alice is sending would contain some session negotiation parameters that cou=
ld be taken to imply what sorts of media types Alice wants to allow or forb=
id in the conference. But I will reiterate that RFC4579 and the related spe=
cifications I can think of off the top of my head do not contain a provisio=
n for the INVITE to communicate session policy information - and I don't th=
ink it is safe to infer from the SDP in an INVITE that Alice intends to lim=
it the conference to the media type(s) it offers.

Moreover, in the ad hoc conference corner case, I see no plausible justific=
ation for why the conference bridge would ever reject Alice's initial INVIT=
E based on the media type it contains. If she's sending her INVITE through =
the IdP via SIP, it would never even forward the request, let alone issue a=
 token for it, if it didn't like the media type that Alice was proposing - =
if Alice proposes "video", and the IdP only allows "audio", when the IdP re=
ceives Alice's INVITE containing "video" it would do the rejecting itself (=
in SIP or HTTP). So once again, I don't see how this token would ever lead =
to the actual INVITE that carried the token ever being rejected by the conf=
erence bridge based on the media types authorized by the token.

Jon Peterson
Neustar, Inc.


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From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Date: Thu, 23 Jun 2016 10:21:04 -0400
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To: "Peterson, Jon" <jon.peterson@neustar.biz>
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Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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On Wed, Jun 22, 2016 at 6:29 PM, Peterson, Jon <jon.peterson@neustar.biz>
wrote:

>
> The token added to the INVITE is *not a policy*; it is an authorization
> by the AuthZ Server to allow this request to create a conference focus.
>
>
> Well, this is the complexity level I was trying to avoid, but, I do agree
> that what your proposed token carries isn't strictly a conference policy =
-
> it is instead a policy that constrains conference policies that users are
> allowed to set.
>
> The thing that RFC4353 calls "conference policy" is essentially a policy
> for a particular instance of a conference. The two attributes you initial=
ly
> proposed actually constrain conference policies rather than conferences -
> that is, they apply to all conferences that I might try to set policy for=
,
> rather than any given conference instance. When I set up a particular
> conference, for example, I might want it to be voice only, and to have fo=
ur
> seats. That's the policy I want for that conference. I gather the
> attributes you proposed are the constraints that my local administrative
> domain imposes on conferences I create, so the conference policy I set fo=
r
> any given policy needs to be a subset of those constraints. If my
> administrative domain limits me to only three-seat conferences, then when=
 I
> try to set a conference policy for four seats, that manipulation of
> conference policy would presumably be rejected when it is compared to my
> administrative domain's constraints.
>
> But of course, in the ad hoc corner case you are considering, the
> conference bridge has no way of anticipating how many users Alice will tr=
y
> to add to the conference - so, her initial INVITE containing the paramete=
r
> would never be rejected for violating the constraint on conference size,
> only later INVITEs would be. That attribute has nothing to do with
> authorizing the INVITE carrying the token.
>

Let me give you an example from the Web that I use all the time.
I use Google Drive to share files with others; to do that I go to specific
file/directory that I would like to share and add the email of the person
that I would like to share the file with, and get a link to that
file/directory.
I send that link to the person that owns the email address, which will be
able to use his email address and his credentials to get access to the
file/directory; the system will prevent anybody else from accessing that
file.

So, Google Drive has no way of knowing that someone else other than the one
that I authorized would try to access the file, but it has a clear
understanding of the authorization around this specific file/directory that
it enforces for any access attempts that comes after I set the
authorization.

In that sense, it is similar to this conference use case.

Why is this a problem? what is it about the conference server enforcing a
specific authorization on later requests that is problematic?



>
> Moreover, how does one actually manipulate my conference policy? Accordin=
g
> to RFC4353, it isn't with SIP. Probably with the web, with an HTTP reques=
t.
> RFC4579 et al do not provide provisions for setting conference policy.
> Adding the attributes you proposed to SIP isn't really about setting
> conference policy, it's about alerting the conference service to the
> constraints on conference policy that are imposed by my administrative
> domain. This is an important distinction, because it shows how far remove=
d
> that SIP transaction actually is from the way conference policy is set. W=
hy
> would I send a SIP request with the constraints on conference policy if I=
'm
> only going to have to use HTTP to actually set conference policy later?
>

Sorry, but this last sentence is not clear to me. What do you mean by "f
I'm only going to have to use HTTP to actually set conference policy later?=
"
Remember that the conference policy is now managed on the AuthNZ server,
not the conference server.



> Why wouldn't my administrative domain have me attach its imposed
> constraints on conference policy to the actual HTTP message that is
> manipulating conference policy?
>
> <snip>
>
>
> Now, Alice decides to start an audio conference; to do that, the phone
> obtains an access token from the AuthNZ server that allows Alice to start=
 a
> conference of the type =E2=80=9C*audio=E2=80=9D*.
>
>
> I did figure you had something like this is mind - perhaps that Alice
> would send her INVITE to the IdP first, and that the IdP would then
>  inspect and authorize a conference policy that Alice somehow conveyed vi=
a
> the INVITE. Through what field or parameter do you suppose to INVITE will
> convey that Alice intends for the conference to be restricted to audio
> media? Or do you perhaps mean that Alice would use HTTP to inform her IdP
> of the media choices for the conference - in the which case policy for th=
e
> conference starts in the HTTP realm, and should certainly stay there.
>

No, I am not suggesting to use SIP to communicate with the IdP at all. The
idea is to utilize existing mechanisms that know how to do that already.
In our OAuth draft, we proposed to use HTTP to communicate with the IdP and
SIP to carry the token(s) or code.
https://datatracker.ietf.org/doc/draft-yusef-sipcore-sip-oauth/



>
> I imagine that in the corner case of an ad hoc conference, the INVITE tha=
t
> Alice is sending would contain some session negotiation parameters that
> could be taken to imply what sorts of media types Alice wants to allow or
> forbid in the conference. But I will reiterate that RFC4579 and the relat=
ed
> specifications I can think of off the top of my head do not contain a
> provision for the INVITE to communicate session policy information - and =
I
> don't think it is safe to infer from the SDP in an INVITE that Alice
> intends to limit the conference to the media type(s) it offers.
>
> Moreover, in the ad hoc conference corner case, I see no plausible
> justification for why the conference bridge would ever reject Alice's
> initial INVITE based on the media type it contains. If she's sending her
> INVITE through the IdP via SIP, it would never even forward the request,
> let alone issue a token for it, if it didn't like the media type that Ali=
ce
> was proposing - if Alice proposes "video", and the IdP only allows "audio=
",
> when the IdP receives Alice's INVITE containing "video" it would do the
> rejecting itself (in SIP or HTTP).
>

Yes, that is the whole idea; to delegate the authorization part from the
conference server; the only thing that the conference will be doing now is
enforcing the authorization provided in the token, assuming the token is
valid and provided by a trusted entity.



> So once again, I don't see how this token would ever lead to the actual
> INVITE that carried the token ever being rejected by the conference bridg=
e
> based on the media types authorized by the token.
>

So what? why is this a problem?

Regards,
 Rifaat



>
> Jon Peterson
> Neustar, Inc.
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Wed, Jun 22, 2016 at 6:29 PM, Peterson, Jon <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:jon.peterson@neustar.biz" target=3D"_blank">jon.peterson@ne=
ustar.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-style:solid;=
border-left-color:rgb(204,204,204);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 class=3D"">
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div>
<div>
<div dir=3D"ltr">
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:115%"><font face=3D"monospace,monospace">The token a=
dded to the INVITE is
<b>not a policy</b>; it is an authorization by the AuthZ Server to allow th=
is request to create a conference focus.</font></span></p>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span><div>Well, this is the complexity level I was trying to avoid, but, =
I do agree that what your proposed token carries isn&#39;t strictly a confe=
rence policy - it is instead a policy that constrains conference policies t=
hat users are allowed to set.</div>
<div><br>
</div>
<div>The thing that RFC4353 calls &quot;conference policy&quot; is essentia=
lly a policy for a particular instance of a conference. The two attributes =
you initially proposed actually constrain conference policies rather than c=
onferences - that is, they apply to all conferences
 that I might try to set policy for, rather than any given conference insta=
nce. When I set up a particular conference, for example, I might want it to=
 be voice only, and to have four seats. That&#39;s the policy I want for th=
at conference. I gather the attributes
 you proposed are the constraints that my local administrative domain impos=
es on conferences I create, so the conference policy I set for any given po=
licy needs to be a subset of those constraints. If my administrative domain=
 limits me to only three-seat conferences,
 then when I try to set a conference policy for four seats, that manipulati=
on of conference policy would presumably be rejected when it is compared to=
 my administrative domain&#39;s constraints.=C2=A0</div>
<div><br>
</div>
<div>But of course, in the ad hoc corner case you are considering, the conf=
erence bridge has no way of anticipating how many users Alice will try to a=
dd to the conference - so, her initial INVITE containing the parameter woul=
d never be rejected for violating
 the constraint on conference size, only later INVITEs would be. That attri=
bute has nothing to do with authorizing the INVITE carrying the token.</div=
></div></blockquote><div><br></div><div>Let me give you an example from the=
 Web that I use all the time.<br></div><div>I use Google Drive to share fil=
es with others; to do that I go to specific file/directory that I would lik=
e to share and add the email of the person that I would like to share the f=
ile with, and get a link to that file/directory.</div><div>I send that link=
 to the person that owns the email address, which will be able to use his e=
mail address and his credentials to get access to the file/directory; the s=
ystem will prevent anybody else from accessing that file.</div><div><br></d=
iv><div>So, Google Drive has no way of knowing that someone else other than=
 the one that I authorized would try to access the file, but it has a clear=
 understanding of the authorization around this specific file/directory tha=
t it enforces for any access attempts that comes after I set the authorizat=
ion.</div><div><br></div><div>In that sense, it is similar to this conferen=
ce use case.=C2=A0</div><div><br></div><div>Why is this a problem? what is =
it about the conference server enforcing a specific authorization on later =
requests that is problematic?<br></div><div><br></div><div>=C2=A0</div><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left=
-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);paddi=
ng-left:1ex"><div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:=
14px;font-family:Calibri,sans-serif">
<div><br>
</div>
<div>Moreover, how does one actually manipulate my conference policy? Accor=
ding to RFC4353, it isn&#39;t with SIP. Probably with the web, with an HTTP=
 request. RFC4579 et al do not provide provisions for setting conference po=
licy. Adding the attributes you proposed
 to SIP isn&#39;t really about setting conference policy, it&#39;s about al=
erting the conference service to the constraints on conference policy that =
are imposed by my administrative domain. This is an important distinction, =
because it shows how far removed that SIP
 transaction actually is from the way conference policy is set. Why would I=
 send a SIP request with the constraints on conference policy if I&#39;m on=
ly going to have to use HTTP to actually set conference policy later? </div=
></div></blockquote><div><br></div><div>Sorry, but this last sentence is no=
t clear to me. What do you mean by &quot;<span style=3D"color:rgb(0,0,0);fo=
nt-family:Calibri,sans-serif;font-size:14px">f I&#39;m only going to have t=
o use HTTP to actually set conference policy later?</span>&quot;</div><div>=
Remember that the conference policy is now managed on the AuthNZ server, no=
t the conference server.</div><div><br></div><div>=C2=A0</div><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1p=
x;border-left-style:solid;border-left-color:rgb(204,204,204);padding-left:1=
ex"><div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font=
-family:Calibri,sans-serif"><div>Why wouldn&#39;t my administrative domain =
have
 me attach its imposed constraints on conference policy to the actual HTTP =
message that is manipulating conference policy?</div>
<div><br>
</div>
<div>&lt;snip&gt;</div><span class=3D"">
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<p class=3D"MsoNormal" style=3D"margin-bottom:0.0001pt"><span style=3D"font=
-size:10pt;line-height:15.3333px"><font face=3D"monospace,monospace"><br>
Now, Alice decides to start an audio conference; to do that, the phone obta=
ins an access token from the AuthNZ server that allows Alice to start a con=
ference of the type =E2=80=9C<b>audio=E2=80=9D</b>.</font></span></p>
</div>
</blockquote>
</span>
<div><br>
</div>
</span><div>I did figure you had something like this is mind - perhaps that=
 Alice would send her INVITE to the IdP first, and that the IdP would then =
=C2=A0inspect and authorize a conference policy that Alice somehow conveyed=
 via the INVITE. Through what field or parameter
 do you suppose to INVITE will convey that Alice intends for the conference=
 to be restricted to audio media? Or do you perhaps mean that Alice would u=
se HTTP to inform her IdP of the media choices for the conference - in the =
which case policy for the conference
 starts in the HTTP realm, and should certainly stay there.</div></div></bl=
ockquote><div><br></div><div>No, I am not suggesting to use SIP to communic=
ate with the IdP at all. The idea is to utilize existing mechanisms that kn=
ow how to do that already.</div><div>In our OAuth draft, we proposed to use=
 HTTP to communicate with the IdP and SIP to carry the token(s) or code.</d=
iv><div><a href=3D"https://datatracker.ietf.org/doc/draft-yusef-sipcore-sip=
-oauth/">https://datatracker.ietf.org/doc/draft-yusef-sipcore-sip-oauth/</a=
><br></div><div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote=
" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style=
:solid;border-left-color:rgb(204,204,204);padding-left:1ex"><div style=3D"w=
ord-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-family:Calibri,san=
s-serif">
<div><br>
</div>
<div>I imagine that in the corner case of an ad hoc conference, the INVITE =
that Alice is sending would contain some session negotiation parameters tha=
t could be taken to imply what sorts of media types Alice wants to allow or=
 forbid in the conference. But I
 will reiterate that RFC4579 and the related specifications I can think of =
off the top of my head do not contain a provision for the INVITE to communi=
cate session policy information - and I don&#39;t think it is safe to infer=
 from the SDP in an INVITE that Alice
 intends to limit the conference to the media type(s) it offers.</div>
<div><br>
</div>
<div>Moreover, in the ad hoc conference corner case, I see no plausible jus=
tification for why the conference bridge would ever reject Alice&#39;s init=
ial INVITE based on the media type it contains. If she&#39;s sending her IN=
VITE through the IdP via SIP, it would never
 even forward the request, let alone issue a token for it, if it didn&#39;t=
 like the media type that Alice was proposing - if Alice proposes &quot;vid=
eo&quot;, and the IdP only allows &quot;audio&quot;, when the IdP receives =
Alice&#39;s INVITE containing &quot;video&quot; it would do the rejecting
 itself (in SIP or HTTP). </div></div></blockquote><div><br></div><div><div=
>Yes, that is the whole idea; to delegate the authorization part from the c=
onference server; the only thing that the conference will be doing now is e=
nforcing the authorization provided in the token, assuming the token is val=
id and provided by a trusted entity.</div></div><div><br></div><div>=C2=A0<=
/div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;bo=
rder-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,2=
04);padding-left:1ex"><div style=3D"word-wrap:break-word;color:rgb(0,0,0);f=
ont-size:14px;font-family:Calibri,sans-serif"><div>So once again, I don&#39=
;t see how this token would ever lead to the actual INVITE that carried the=
 token ever being rejected by the conference bridge based on the media type=
s authorized by the token.</div></div></blockquote><div><br></div><div>So w=
hat? why is this a problem?</div><div><br></div><div>Regards,</div><div>=C2=
=A0Rifaat</div><div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_q=
uote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-s=
tyle:solid;border-left-color:rgb(204,204,204);padding-left:1ex"><div style=
=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-family:Calibr=
i,sans-serif"><span class=3D""><font color=3D"#888888">
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
</font></span></div>

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

--001a113d0a84b3ce390535f2c38f--


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From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Date: Thu, 23 Jun 2016 10:32:25 -0400
Message-ID: <CAGL6epLyQJhGvgu=23YfH1s1jQOeY47wGqNTZVTqQVLTERhCuA@mail.gmail.com>
To: Roman Shpount <roman@telurix.com>
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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Roman,

We are specifically talking about an RFC4579 based conference use case,
just to drive the point and discuss the concept and idea of delegating
AuthNZ to other entities.

Regards,
 Rifaat


On Wed, Jun 22, 2016 at 5:02 PM, Roman Shpount <roman@telurix.com> wrote:

> Rifaat,
>
> In your example, I would expect to issue an HTTP request to create the
> conference, get the SIP URI back (with authentication token embedded in the
> URI) and place the SIP call to this URI. I do not see a reason why
> any standardization effort is required to implement this.
>
> Regards,
>
> _____________
> Roman Shpount
>
> On Wed, Jun 22, 2016 at 8:48 AM, Rifaat Shekh-Yusef <rifaat.ietf@gmail.com
> > wrote:
>
>> Roman,
>>
>> I think that the question of how to exactly carry the token in SIP
>> request is a minor issue *at this stage*.
>> We are discussing the idea and concept of delegating both the AuthN and
>> AuthZ to separate server(s) and how to achieve that.
>>
>> Regards,
>>  Rifaat
>>
>>
>> On Tue, Jun 21, 2016 at 7:50 PM, Roman Shpount <roman@telurix.com> wrote:
>>
>>> Furthermore, reading all of this, I cannot understand why Authorization
>>> token cannot be a part of SIP URL, either as user part or URL parameter.
>>> This does not need to be anything more then service specific URL and does
>>> not require a new SIP header or authentication mechanism.
>>>
>>> _____________
>>> Roman Shpount
>>>
>>> On Tue, Jun 21, 2016 at 7:46 PM, Peterson, Jon <jon.peterson@neustar.biz
>>> > wrote:
>>>
>>>>
>>>> It's not really what I'm looking for, no, and after mulling this over a
>>>> bit, I fear going farther along this path would only drill down into levels
>>>> of complexity that will likely make this harder to understand rather than
>>>> easier.
>>>>
>>>> I do acknowledge that you describe the INVITE as being rejected if it
>>>> doesn't contain a token - but that's not the same thing as saying the
>>>> INVITE will be rejected because the relying party makes an authorization
>>>> decision based on the attributes in the token. It's more like an INVITE
>>>> being rejected because a user agent can't supply an Authorization header. I
>>>> will reiterate that I don't think the kinds of attributes you are proposing
>>>> should be in the scope of an effort that will try to address a framework
>>>> for authorizing SIP requests. I don't think the values of the attributes
>>>> (as opposed to the mere presence of some attributes) have anything to do
>>>> with authorizing the SIP request that you propose would carry the
>>>> attributes.
>>>>
>>>> With regards to this whole conferencing example, it does seem a bit
>>>> contrived to me, like you're sketching a SIP-based way of setting
>>>> conference policy on the fly that happens to require the kind of attributes
>>>> you want, and then saying that demonstrates the need for the attributes.
>>>> The fact that all of the ways to do conference policy we can point to in
>>>> fact do it via the web makes it hard for me to understand why we should
>>>> create a SIP-based way of doing this. To me, it looks like you're opening a
>>>> problem space that has much more to do with reshaping the way that SIP does
>>>> conferences (and again, as Radhika's mail pointing out, there's been a lot
>>>> of thinking about this since the days of RFC4579) than it does with
>>>> defining a framework for authorizing SIP requests.
>>>>
>>>> Jon Peterson
>>>> Neustar, Inc.
>>>>
>>>> From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
>>>> Date: Thursday, June 16, 2016 at 6:19 AM
>>>> To: Jon Peterson <jon.peterson@neustar.biz>
>>>> Cc: "sipcore@ietf.org" <sipcore@ietf.org>
>>>> Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
>>>>
>>>> The attributes I provided are examples of attributes that might be
>>>> provided in an access token; it was not meant to include all the attributes
>>>> for this use case, because my intention was not to focus on conference
>>>> server, but to discuss the idea in general.
>>>>
>>>> The following could be used with the conference factory to request a
>>>> conference focus:
>>>> The INVITE sent to the conference factory could include a token that
>>>> has a * conference-type* attribute which could take one of the
>>>> following values: *audio, video, webconference, audio-webconference,
>>>> or video-webconference*.
>>>> The conference factory will then issue a *conference uri* for the
>>>> requested conference type, assuming the token is valid. If the
>>>> *conference-type* attribute is missing, then the request would be
>>>> rejected.
>>>>
>>>> Is that what you are looking for?
>>>>
>>>> Regards,
>>>>  Rifaat
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Wed, Jun 15, 2016 at 5:06 PM, Peterson, Jon <
>>>> jon.peterson@neustar.biz> wrote:
>>>>
>>>>>
>>>>>>
>>>>>> In that case, if you could furnish a use case with attributes that
>>>>>> would be carried in a SIP iNVITE that are actually salient to the
>>>>>> authorization of that particular SIP request, we'd be getting a lot closer
>>>>>> to common ground.
>>>>>>
>>>>>>
>>>>> An obvious example would be an INVITE with a token sent to a
>>>>> conference factory to authorize the creation of a focus.
>>>>>
>>>>>
>>>>> We're still talking past each other. My point above was that the two
>>>>> attributes you specified - constraints on conference size and media types -
>>>>> have nothing to do with the authorization of the INVITE carrying those
>>>>> attributes. Like, unless the conference size permitted is 0, then the
>>>>> INVITE that creates an ad hoc conference and connects you to its bridge is
>>>>> not going to be rejected by the focus on the basis of that conference size
>>>>> attribute. If the maximum conference size were 0, the AuthZ service just
>>>>> wouldn't attach a token to an INVITE as it would never be allowed to start
>>>>> a conference. Same goes for media types.
>>>>>
>>>>> Instead, what your two attributes constrain is what happens after the
>>>>> conference is created: when ten users try to dial in to a conference with a
>>>>> permitted size of nine, say. That tenth INVITE would get rejected - not the
>>>>> conference-forming INVITE that you propose would carry the token. That's
>>>>> why what you're proposing is overloading the INVITE to tunnel conference
>>>>> policy constraints to the focus. I mean, maybe that's a problem space worth
>>>>> investigating, though I suspect there are good reasons why RFC4353
>>>>> specified that conference policy is not manipulated with SIP - mostly
>>>>> because the semantics of manipulating conference policy probably don't map
>>>>> well onto dialog-forming SIP requests. But I think that conference policy
>>>>> provisioning problem space would be pretty far from where we started in
>>>>> this discussion.
>>>>>
>>>>> Let me reiterate what I wrote in my previous mail, which I quote here
>>>>> at the top: "furnish a use case with attributes that would be carried in a
>>>>> SIP iNVITE that are actually salient to the authorization of that
>>>>> particular SIP request." Again: "that particular SIP request." Not later
>>>>> transactions in some broader context in which the original user agent
>>>>> probably isn't even an endpoint.
>>>>>
>>>>> The Authorization header in SIP, for example, is actually used by the
>>>>> UAS to make a decision about whether or not to authorize the request in
>>>>> which the header appears.  The token that you propose a SIP INVITE would
>>>>> carry in this case is not used by the relying party to make a decision
>>>>> about whether or not that INVITE is authorized. That's why I believe this
>>>>> conference policy issue is a different problem space entirely from figuring
>>>>> out how to implement attribute-based authorization for SIP requests.
>>>>>
>>>>> Jon Peterson
>>>>> Neustar, Inc.
>>>>>
>>>>>
>>>>> Regards,
>>>>>  Rifaat
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> sipcore mailing list
>>>> sipcore@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>>
>>>>
>>>
>>
>

--001a113d0632522c920535f2ec3a
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Roman,<div><br></div><div>We are specifically talking abou=
t an RFC4579 based conference use case, just to drive the point and discuss=
 the concept and idea of delegating AuthNZ to other entities.</div><div><br=
></div><div>Regards,</div><div>=C2=A0Rifaat</div><div><br></div></div><div =
class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Wed, Jun 22, 2016 a=
t 5:02 PM, Roman Shpount <span dir=3D"ltr">&lt;<a href=3D"mailto:roman@telu=
rix.com" target=3D"_blank">roman@telurix.com</a>&gt;</span> wrote:<br><bloc=
kquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #cc=
c solid;padding-left:1ex"><div dir=3D"ltr"><span style=3D"color:rgb(0,0,0);=
font-size:12.8px">Rifaat,</span><br><div><span style=3D"color:rgb(0,0,0);fo=
nt-size:12.8px"><br></span></div><div><font color=3D"#000000"><span style=
=3D"font-size:12.8px">In your example, I would expect to issue an HTTP requ=
est to create the conference, get the SIP URI back (with authentication tok=
en embedded in the URI) and place the SIP call to this URI. I do not see a =
reason why any=C2=A0standardization=C2=A0effort is required to implement th=
is.</span></font></div><div><font color=3D"#000000"><span style=3D"font-siz=
e:12.8px"><br></span></font></div><div><font color=3D"#000000"><span style=
=3D"font-size:12.8px">Regards,</span></font></div></div><div class=3D"gmail=
_extra"><br clear=3D"all"><div><div data-smartmail=3D"gmail_signature">____=
_________<span class=3D"HOEnZb"><font color=3D"#888888"><br>Roman Shpount</=
font></span></div></div><div><div class=3D"h5">
<br><div class=3D"gmail_quote">On Wed, Jun 22, 2016 at 8:48 AM, Rifaat Shek=
h-Yusef <span dir=3D"ltr">&lt;<a href=3D"mailto:rifaat.ietf@gmail.com" targ=
et=3D"_blank">rifaat.ietf@gmail.com</a>&gt;</span> wrote:<br><blockquote cl=
ass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;p=
adding-left:1ex"><div dir=3D"ltr">Roman,<div><br></div><div>I think that th=
e question of how to exactly carry the token in SIP request is a minor issu=
e <b>at this stage</b>.<br></div><div>We are discussing the idea and concep=
t of delegating both the AuthN and AuthZ to separate server(s) and how to a=
chieve that.<br></div><div><br></div><div>Regards,</div><div>=C2=A0Rifaat</=
div><div><br></div></div><div><div><div class=3D"gmail_extra"><br><div clas=
s=3D"gmail_quote">On Tue, Jun 21, 2016 at 7:50 PM, Roman Shpount <span dir=
=3D"ltr">&lt;<a href=3D"mailto:roman@telurix.com" target=3D"_blank">roman@t=
elurix.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 dir=
=3D"ltr">Furthermore, reading all of this, I cannot understand why Authoriz=
ation token cannot be a part of SIP URL, either as user part or URL paramet=
er. This does not need to be anything more then service specific URL and do=
es not require a new SIP header or authentication mechanism.</div><div clas=
s=3D"gmail_extra"><br clear=3D"all"><div><div data-smartmail=3D"gmail_signa=
ture">_____________<br>Roman Shpount</div></div>
<br><div class=3D"gmail_quote"><div><div>On Tue, Jun 21, 2016 at 7:46 PM, P=
eterson, Jon <span dir=3D"ltr">&lt;<a href=3D"mailto:jon.peterson@neustar.b=
iz" target=3D"_blank">jon.peterson@neustar.biz</a>&gt;</span> wrote:<br></d=
iv></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;borde=
r-left:1px #ccc solid;padding-left:1ex"><div><div>



<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>It&#39;s not really what I&#39;m looking for, no, and after mulling th=
is over a bit, I fear going farther along this path would only drill down i=
nto levels of complexity that will likely make this harder to understand ra=
ther than easier.</div>
<div><br>
</div>
<div>I do acknowledge that you describe the INVITE as being rejected if it =
doesn&#39;t contain a token - but that&#39;s not the same thing as saying t=
he INVITE will be rejected because the relying party makes an authorization=
 decision based on the attributes in the
 token. It&#39;s more like an INVITE being rejected because a user agent ca=
n&#39;t supply an Authorization header. I will reiterate that I don&#39;t t=
hink the kinds of attributes you are proposing should be in the scope of an=
 effort that will try to address a framework
 for authorizing SIP requests. I don&#39;t think the values of the attribut=
es (as opposed to the mere presence of some attributes) have anything to do=
 with authorizing the SIP request that you propose would carry the attribut=
es.</div>
<div><br>
</div>
<div>
<div>With regards to this whole conferencing example, it does seem a bit co=
ntrived to me, like you&#39;re sketching a SIP-based way of setting confere=
nce policy on the fly that happens to require the kind of attributes you wa=
nt, and then saying that demonstrates
 the need for the attributes. The fact that all of the ways to do conferenc=
e policy we can point to in fact do it via the web makes it hard for me to =
understand why we should create a SIP-based way of doing this. To me, it lo=
oks like you&#39;re opening a problem
 space that has much more to do with reshaping the way that SIP does confer=
ences (and again, as Radhika&#39;s mail pointing out, there&#39;s been a lo=
t of thinking about this since the days of RFC4579) than it does with defin=
ing a framework for authorizing SIP requests.</div>
</div>
<div><br>
</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-BOTTOM:medium none;BORDER-LEFT:medium none;PADDING-BOTTOM:0in;PADD=
ING-LEFT:0in;PADDING-RIGHT:0in;BORDER-TOP:#b5c4df 1pt solid;BORDER-RIGHT:me=
dium 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" target=3D"_blank">rifaat.ietf@gmail.com<=
/a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Thursday, June 16, 2016 at 6:=
19 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>&quot;<a href=3D"mailto:sipcore=
@ietf.org" target=3D"_blank">sipcore@ietf.org</a>&quot; &lt;<a href=3D"mail=
to:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [sipcore] SIP AuthNZ P=
roblem Statement - v3<br>
</div>
<div><br>
</div>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div>
<div>
<div dir=3D"ltr">
<div>The attributes I provided are examples of attributes that might be pro=
vided in an access token; it was not meant to include all the attributes fo=
r this use case, because my intention was not to focus on conference server=
, but to discuss the idea in general.</div>
<div><br>
</div>
<div>
<div>The following could be used with the conference factory to request a c=
onference focus:</div>
<div>The INVITE sent to the conference factory could include a token that h=
as a <b>
conference-type</b> attribute which could take one of the following values:=
 <b>audio, video, webconference, audio-webconference, or video-webconferenc=
e</b>.</div>
<div>The conference factory will then issue a <b>conference uri</b> for the=
 requested conference type, assuming the token is valid. If the
<b>conference-type</b> attribute is missing, then the request would be reje=
cted.</div>
</div>
<div><br>
</div>
<div>Is that what you are looking for?</div>
<div><br>
</div>
<div>Regards,</div>
<div>=C2=A0Rifaat</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
</div>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Wed, Jun 15, 2016 at 5:06 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:0 0 0 .8ex;border-left:1p=
x #ccc 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><span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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">
<span><br>
<br>
</span>
<div>
<div>In that case, if you could furnish a use case with attributes that wou=
ld be carried in a SIP iNVITE that are actually salient to the authorizatio=
n of that particular SIP request, we&#39;d be getting a lot closer to commo=
n ground.</div>
<div><br>
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>An obvious example would be an INVITE with a token sent to a conferenc=
e factory to authorize the creation of a focus.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span>
<div>We&#39;re still talking past each other. My point above was that the t=
wo attributes you specified - constraints on conference size and media type=
s - have nothing to do with the authorization of the INVITE carrying those =
attributes. Like, unless the conference
 size permitted is 0, then the INVITE that creates an ad hoc conference and=
 connects you to its bridge is not going to be rejected by the focus on the=
 basis of that conference size attribute. If the maximum conference size we=
re 0, the AuthZ service just wouldn&#39;t
 attach a token to an INVITE as it would never be allowed to start a confer=
ence. Same goes for media types.</div>
<div><br>
</div>
<div>Instead, what your two attributes constrain is what happens after the =
conference is created: when ten users try to dial in to a conference with a=
 permitted size of nine, say. That tenth INVITE would get rejected - not th=
e conference-forming INVITE that
 you propose would carry the token. That&#39;s why what you&#39;re proposin=
g is overloading the INVITE to tunnel conference policy constraints to the =
focus. I mean, maybe that&#39;s a problem space worth investigating, though=
 I suspect there are good reasons why RFC4353
 specified that conference policy is not manipulated with SIP - mostly beca=
use the semantics of manipulating conference policy probably don&#39;t map =
well onto dialog-forming SIP requests. But I think that conference policy p=
rovisioning problem space would be pretty
 far from where we started in this discussion.</div>
<div><br>
</div>
<div>Let me reiterate what I wrote in my previous mail, which I quote here =
at the top: &quot;furnish a use case with attributes that would be carried =
in a SIP iNVITE that are actually salient to the authorization of that part=
icular SIP request.&quot; Again: &quot;that particular
 SIP request.&quot; Not later transactions in some broader context in which=
 the original user agent probably isn&#39;t even an endpoint.</div>
<div><br>
</div>
<div>The Authorization header in SIP, for example, is actually used by the =
UAS to make a decision about whether or not to authorize the request in whi=
ch the header appears.=C2=A0 The token that you propose a SIP INVITE would =
carry in this case is not used by the
 relying party to make a decision about whether or not that INVITE is autho=
rized. That&#39;s why I believe this conference policy issue is a different=
 problem space entirely from figuring out how to implement attribute-based =
authorization for SIP requests.</div>
<span>
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<div>Regards,</div>
<div>=C2=A0Rifaat</div>
<div><br>
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Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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On Thu, Jun 23, 2016 at 10:32 AM, Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
wrote:

> Roman,
>
> We are specifically talking about an RFC4579 based conference use case,
> just to drive the point and discuss the concept and idea of delegating
> AuthNZ to other entities.
>
>
I think it is more like we are discussing spherical elephants in vacuum. I
am not aware of anybody using  RFC4579 based conference in such a manner or
ever needing to so.

Is there a practical application for this framework?
_____________
Roman Shpount

--001a113ed0d01ceade0535f309eb
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure" data-smartmail=3D"gmail_signature">On Thu, Jun 23, 2016 at 10:32 AM, R=
ifaat Shekh-Yusef <span dir=3D"ltr">&lt;<a href=3D"mailto:rifaat.ietf@gmail=
.com" target=3D"_blank">rifaat.ietf@gmail.com</a>&gt;</span> wrote:<br></di=
v></div><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=
=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding=
-left:1ex"><div dir=3D"ltr">Roman,<div><br></div><div>We are specifically t=
alking about an RFC4579 based conference use case, just to drive the point =
and discuss the concept and idea of delegating AuthNZ to other entities.</d=
iv><div><br></div></div></blockquote><div><br></div><div>I think it is more=
 like we are discussing spherical elephants in vacuum. I am not aware of an=
ybody using=C2=A0=C2=A0RFC4579 based conference in such a manner or ever ne=
eding to so.</div><div><br></div><div>Is there a practical application for =
this framework?</div><div><div class=3D"gmail_signature" data-smartmail=3D"=
gmail_signature">_____________<br>Roman Shpount</div></div><div>=C2=A0</div=
></div></div></div>

--001a113ed0d01ceade0535f309eb--


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From nobody Thu Jun 23 12:04:35 2016
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From: worley@ariadne.com (Dale R. Worley)
To: "Ben Campbell" <ben@nostrum.com>
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Subject: Re: [sipcore] AD Evaluation of draft-ietf-sipcore-dns-dual-stack-06
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"Ben Campbell" <ben@nostrum.com> writes:
> -1, 2nd paragraph, first sentence:
>
> That sentence is a bit hard to parse, please consider breaking into two 
> or more simpler sentences.

Broken into two sentences.  Yes, the joining of the two major clauses of
that sentence wasn't done quite correctly.

> -2, definition of IPv4/IPv6 UA/UAC/UAS:
>
> This defines one term, then declares the draft will use another. Why not 
> just define _that_ term in the first place?

Done.

> -4:
>
> This section kind of buries the lede, and leaves me wondering what the 
> actual clarification was. Is the point to say not to interleave the 
> results of the address record queries among srv records? Is there a 
> concern that people thought they were to do something different? It 
> would be helpful to say what misconception you intend to clarify.

I've revised the first paragraph; it now states a MUST for the important
point and a MUST NOT for what we have been fearing people might do.

> -5, first paragraph, last sentence:
>
> Oddly constructed sentence. The structure seems to imply a tension 
> between the idea of optimizing performance and not introducing security 
> considerations. Please consider something to the effect of the 
> following:
>
> "Both of these procedures are optimization designed to improve the 
> performance of dual-stack clients. Neither introduce new security 
> considerations."

I've reorganized the paragraph.  And now it states the presupposition
that underlies the conclusion that no new security considerations are
introduced:  It's assumed that any address that can be derived from a
URI might be contacted to send a message to that URI.  E.g., security
does not depend on SRV priority always being followed.

Dale


From nobody Thu Jun 23 12:47:29 2016
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From: "Ben Campbell" <ben@nostrum.com>
To: "Dale R. Worley" <worley@ariadne.com>
Date: Thu, 23 Jun 2016 14:47:18 -0500
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Subject: Re: [sipcore] AD Evaluation of draft-ietf-sipcore-dns-dual-stack-06
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It all sounds good (pending actual text).

Thanks!

Ben.

On 23 Jun 2016, at 14:04, Dale R. Worley wrote:

> "Ben Campbell" <ben@nostrum.com> writes:
>> -1, 2nd paragraph, first sentence:
>>
>> That sentence is a bit hard to parse, please consider breaking into two
>> or more simpler sentences.
>
> Broken into two sentences.  Yes, the joining of the two major clauses of
> that sentence wasn't done quite correctly.
>
>> -2, definition of IPv4/IPv6 UA/UAC/UAS:
>>
>> This defines one term, then declares the draft will use another. Why not
>> just define _that_ term in the first place?
>
> Done.
>
>> -4:
>>
>> This section kind of buries the lede, and leaves me wondering what the
>> actual clarification was. Is the point to say not to interleave the
>> results of the address record queries among srv records? Is there a
>> concern that people thought they were to do something different? It
>> would be helpful to say what misconception you intend to clarify.
>
> I've revised the first paragraph; it now states a MUST for the important
> point and a MUST NOT for what we have been fearing people might do.
>
>> -5, first paragraph, last sentence:
>>
>> Oddly constructed sentence. The structure seems to imply a tension
>> between the idea of optimizing performance and not introducing security
>> considerations. Please consider something to the effect of the
>> following:
>>
>> "Both of these procedures are optimization designed to improve the
>> performance of dual-stack clients. Neither introduce new security
>> considerations."
>
> I've reorganized the paragraph.  And now it states the presupposition
> that underlies the conclusion that no new security considerations are
> introduced:  It's assumed that any address that can be derived from a
> URI might be contacted to send a message to that URI.  E.g., security
> does not depend on SRV priority always being followed.
>
> Dale
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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Subject: [sipcore] Happy Eyeballs:  Targets
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This thread is about solutions for this part of the requirements:

    - Solve the "happy eyeballs" problem for INVITE, i.e., if a
      destination URI is advertised with both multiple targets (IPv4 and
      IPv6 or otherwise), but connectivity to only one target exists,
      usually message transmission will not be delayed unacceptably, and in
      particular, not by a full timeout as prescribed in 3261.

    Note that the solution may be decidedly different based on the
    transport protocol(s) for the targets.

    There is uncertainty about whether adding an RTT to signaling times is
    acceptable.  For example, in practice RTT on the overall Internet is
    less than 1/2 second, and that is an acceptable delay in call setup
    times under ordinary circumstances.

    - The solution must approximate 3263, in that if several targets are
      specified by *different* SRV records and are reachable over a long
      period of time, the relative traffic shares sent to them must be
      compatible with the priorities and weights of the SRV records.  (If
      they are alternatives that are *not* derived from different SRV
      records, the solution is unconstrained regarding relative traffic
      shares.)

Comments are invited!

Useful definitions:

a "client" is the entity that wishes to send a SIP message

a "target" or "transport target" is an address/port/protocol triple
that is an address for the transport layer of the stack.  A target is
derived from a SIP URI (for a request) or a host-port (for a
response).

a "flow" is a sequence of related messages between the client and a
target.  If the protocol is connection-oriented, the flow encompasses
the connection.  If the protocol requires cryptographic setup, the
flow encompasses the cryptographic session.

a "probe" is an operation executed on a flow by a client to determine
whether it can successfully communicate with the target, without
changing the SIP dialog state with the target.  Probes can take many
forms:

    Setting up a connection of a connection-oriented protocol.

    Performing a cryptographic handshake.

    Performing a keep-alive as defined by the protocol.

    Sending an OPTIONS request with Max-Forwards 0.

Note that the last choice can be used with any(!) protocol.  If the
OPTIONS reaches the target, the target is required to respond with
either a 200 or 483 response (without forwarding it to another
entity).  Conveniently, a server can respond to such a request
statelessly, so such requests are low-overhead.

The current state of the solution (as I know of it) is:

Note that some SIP messages are time-sensitive for the usesr
experience (e.g., initial INVITEs), while others are not (e.g., a
refreshing REGISTER).  A client MAY choose not to apply the following
rules for non-time-sensitive messages.

Devices MAY change the target order prescribed by RFC 3263/2782.
  The device SHOULD follow the Happy Eyeballs rules, viz.:
     - Prefer targets with existing flows
     - Prefer targets with a different address family than that of a
         non-responsive address
     - Deprecate targets known to be non-responsive
     - May simultaneously initiate flows with multiple targets as long
         as UA does not have more than one simultaneous outstanding
         copy of a request
     - Prefer simultaneously initiating flows with targets in different
         address families

Devices MAY contact targets in any order, including those obtained via
different SRV records, notwithstanding the priority/weight specified
in the SRV records.  But in doing this, they MUST approximate the
behavior specified by RFC 3263, in this sense:

    If a set of targets is all of the targets derived from two or more
    SRV records, and at least one target for each SRV record is
    reachable by the client over a long period of time, the relative
    shares of traffic sent to each subset of targets derived from one
    SRV record must converge to the traffic shares that would result
    if the client contacted these targets in the order specified by
    RFC 3263 (i.e., according to the priorities and weights of the SRV
    records).

    (Note that the relative traffic shares between targets that are
    *not* derived from different SRV records (e.g., alternative A
    records for a DNS name) are not constrained by this requirement.)

In general, this means that cached reachability information about
targets should time out, causing the behavior of the client to revert
to RFC 3263 over time.

(Beware that we have to define "reachable" above to include
responsiveness -- a high-priority target that has a 5 sec RTT
shouldn't be able to commandeer all of the traffic.)

If a client does not have recent reachability information for the flow
to a given target, the client SHOULD probe the flow before sending a
request to the target.

    This is because in the worst case, sending a request commits the
    client to waiting for a timeout before it can send a duplicate
    request to another target.  Note that probes do not change the SIP
    dialog state of any entity, so probes can be sent in parallel to
    multiple targets.

Reduce client transaction timeouts:
    Timer B and Timer F are currently 64*T1, which defaults to 32 seconds

It seems that reducing the default T1 from 500 msec to 100 msec
suffices for this.  It seems that RTT to arbitrary places on the
Internet can take as long as 500 msec, but RTT to web servers
generally takes 100 msec or less.  That argues for reducing T1 to 100
msec, which makes timers B/F 6.4 sec.  In practice, SIP servers are
likely to have connectivity like web servers.  But we want global
public SIP to work (e.g., in peer-to-peer SIP), so SIP to arbitrary
addresses should only rarely time out.

The retransmission schedule specified by RFC 3261 is:

    1st send is at time 0
    2nd send is at time T1
    3rd send is at time 3*T1
    4th send is at time 7*T1
    5th send is at time 15*T1
    6th send is at time 31*T1
    7th send is at time 63*T1
    timer B fires at time 64*T1, terminating the transmission

This is just long enough to allow 7 transmissions of the message.  If
we reduce T1 to 100 msec (from 500 msec), the total length of the
schedule is reduced to 6.4 sec, which seems tolerable as a fail-over
delay.  It still alllows 7 retransmissions.

Dale


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From: Roman Shpount <roman@telurix.com>
Date: Fri, 24 Jun 2016 00:11:06 -0400
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Dale,

One more mechanism for checking a UDP flows is STUN keep-alive message
described in https://tools.ietf.org/html/rfc5626#section-3.5.2. These
messages are even lower overhead then OPTIONS messages.

Regards,
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra">Dale,</div><div class=3D"gmail_=
extra"><br></div><div class=3D"gmail_extra">One more mechanism for checking=
 a UDP flows is STUN keep-alive message described in=C2=A0<a href=3D"https:=
//tools.ietf.org/html/rfc5626#section-3.5.2">https://tools.ietf.org/html/rf=
c5626#section-3.5.2</a>. These messages are even lower overhead then OPTION=
S messages.</div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_e=
xtra">Regards,<br clear=3D"all"><div><div class=3D"gmail_signature" data-sm=
artmail=3D"gmail_signature">_____________<br>Roman Shpount</div></div>
<br></div></div>

--001a113ed0d01698480535fe5c41--


From nobody Fri Jun 24 09:04:56 2016
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Subject: [sipcore] sipcore - Requested session has been scheduled for IETF 96
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Dear Adam Roach,

The session(s) that you have requested have been scheduled.
Below is the scheduled session information followed by
the original request. 

sipcore Session 1 (1:00:00)
    Wednesday, Afternoon Session II 1550-1720
    Room Name: Charlottenburg I size: 80
    ---------------------------------------------
    


Request Information:


---------------------------------------------------------
Working Group Name: Session Initiation Protocol Core
Area Name: Applications and Real-Time Area
Session Requester: Adam Roach

Number of Sessions: 1
Length of Session(s):  1 Hour
Number of Attendees: 25
Conflicts to Avoid: 
 First Priority: netvc xrblock webpush stir mmusic rtcweb avtext perc avtcore dispatch




Special Requests:
  Cannot conflict with interledger bof.
---------------------------------------------------------


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Hi,

> Reduce client transaction timeouts:
>     Timer B and Timer F are currently 64*T1, which defaults to 32
seconds
>
> It seems that reducing the default T1 from 500 msec to 100 msec
> suffices for this.

The issue with using a smaller T1 value is that it doesn't just impact the
desired timer.  For instance, it can cause more retries.  Because of this,
the draft might want to specify allowing the specific smaller timers in a
way which doesn't rely upon a smaller T1.

If it matters, RFC 3261 section 17.1.1.2 has normative statements
concerning the topic of lowing T1.

Thanks,
Brett


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Moreover, in the ad hoc conference corner case, I see no plausible justific=
ation for why the conference bridge would ever reject Alice's initial INVIT=
E based on the media type it contains. If she's sending her INVITE through =
the IdP via SIP, it would never even forward the request, let alone issue a=
 token for it, if it didn't like the media type that Alice was proposing - =
if Alice proposes "video", and the IdP only allows "audio", when the IdP re=
ceives Alice's INVITE containing "video" it would do the rejecting itself (=
in SIP or HTTP).

Yes, that is the whole idea; to delegate the authorization part from the co=
nference server; the only thing that the conference will be doing now is en=
forcing the authorization provided in the token, assuming the token is vali=
d and provided by a trusted entity.

... um, I would have thought this totally undermined your argument. If the =
IdP simply prevents the INVITE from ever being sent to the conference bridg=
e when the proposed media type is unacceptable to the IdP, then there's no =
need to carry attribute to the conference bridge at all. If the IdP doesn't=
 like the media type, the conference bridge will never see the INVITE; and =
if the conference bridge sees the INVITE, then the IdP must have liked the =
media type. Attribute adds nothing.

This is a consequence of architecting an administrative domain in a draconi=
an way that lets intermediaries prevent user agents from sending SIP messag=
es where they like, of course. But even if we assume a more open environmen=
t where the user agent can send INVITEs directly to the conference bridge, =
all the conference bridge needs is a signature or similar token that indica=
tes the IdP's approval, -not- any specific attribute.


So once again, I don't see how this token would ever lead to the actual INV=
ITE that carried the token ever being rejected by the conference bridge bas=
ed on the media types authorized by the token.

So what? why is this a problem?

It's a problem because you're proposing to take a feature (OAuth) that exis=
ts in HTTP and to add it to SIP.  To make an argument for that, you at leas=
t must establish that SIP needs this feature - and preferably you need to e=
stablish that making this protocol change and performing these operations i=
n SIP is less work (in the grand scheme of implementation and deployment ov=
erall) than saying that if you want to perform operations like this you sho=
uld instead just use HTTP. (Which you can, and for these conference cases, =
you surely should.)

What I postulated way back at the beginning of this thread is that the kind=
 of applications that benefit from attribute-based authorization tend to be=
 controlled by the web rather than by SIP, and that this is why, although w=
e've been studying this problem space for more than a decade, we never stan=
dardized an attribute-based authorization protocol for SIP. This entire con=
ferencing example has been about manipulating conference policy and managin=
g constraints on conference policy, which may be a problem space with unsol=
ved issues (honestly, I'm not much of an expert on the current thinking abo=
ut conferencing) but I don't think it's the SIP authorization problem space=
.

The semantics of the INVITE method are "hey, remote resource, I'd like to s=
et up a session with you involving the following media types." The question=
 of whether or not you accept INVITEs, which is to my mind the SIP authoriz=
ation problem, is mainly about figuring out why you would say yes or no to =
that question. What you are proposing here is a conference policy constrain=
t token that lets the IdP piggyback information on the INVITE to be sent to=
 the conference bring. We've struggled here to even identify how that infor=
mation would alter to the decision about whether or not to accept the INVIT=
E. If we need to standardize a protocol mechanism for that function, I have=
 a hard time seeing why SIP would be our first choice, and then why an INVI=
TE setting up a session would be the right choice, versus just doing this w=
ith a protocol interaction that has a semantics other than "I'd like to set=
 up a session with you."

I'd note that your Google drive example is also a pretty poor fit for these=
 semantics. But these are the primary semantics of SIP. If the thing you wa=
nt to do has different semantics, there's probably a way to do with a diffe=
rent protocol. Albeit, that doesn't mean there isn't an authorization probl=
em for SIP worth identifying and considering.

Jon Peterson
Neustar, Inc.


Regards,
 Rifaat



--_000_D392BAC11A0530jonpetersonneustarbiz_
Content-Type: text/html; charset="us-ascii"
Content-ID: <14E34135CD317746A75ED5FE2127B889@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;">
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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>
<br>
</div>
<div>Moreover, in the ad hoc conference corner case, I see no plausible jus=
tification for why the conference bridge would ever reject Alice's initial =
INVITE based on the media type it contains. If she's sending her INVITE thr=
ough the IdP via SIP, it would never
 even forward the request, let alone issue a token for it, if it didn't lik=
e the media type that Alice was proposing - if Alice proposes &quot;video&q=
uot;, and the IdP only allows &quot;audio&quot;, when the IdP receives Alic=
e's INVITE containing &quot;video&quot; it would do the rejecting
 itself (in SIP or HTTP). </div>
</div>
</blockquote>
<div><br>
</div>
<div>
<div>Yes, that is the whole idea; to delegate the authorization part from t=
he conference server; the only thing that the conference will be doing now =
is enforcing the authorization provided in the token, assuming the token is=
 valid and provided by a trusted
 entity.</div>
</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>... um, I would have thought this totally undermined your argument. If=
 the IdP simply prevents the INVITE from ever being sent to the conference =
bridge when the proposed media type is unacceptable to the IdP, then there'=
s no need to carry attribute to
 the conference bridge at all. If the IdP doesn't like the media type, the =
conference bridge will never see the INVITE; and if the conference bridge s=
ees the INVITE, then the IdP must have liked the media type. Attribute adds=
 nothing.</div>
<div><br>
</div>
<div>This is a consequence of architecting an administrative domain in a dr=
aconian way that lets intermediaries prevent user agents from sending SIP m=
essages where they like, of course. But even if we assume a more open envir=
onment where the user agent can
 send INVITEs directly to the conference bridge, all the conference bridge =
needs is a signature or similar token that indicates the IdP's approval, -n=
ot- any specific attribute.</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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>So once again, I don't see how this token would ever lead to the actua=
l INVITE that carried the token ever being rejected by the conference bridg=
e based on the media types authorized by the token.</div>
</div>
</blockquote>
<div><br>
</div>
<div>So what? why is this a problem?</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>It's a problem because you're proposing to take a feature (OAuth) that=
 exists in HTTP and to add it to SIP. &nbsp;To make an argument for that, y=
ou at least must establish that SIP needs this feature - and preferably you=
 need to establish that making this protocol
 change and performing these operations in SIP is less work (in the grand s=
cheme of implementation and deployment overall) than saying that if you wan=
t to perform operations like this you should instead just use HTTP. (Which =
you can, and for these conference
 cases, you surely should.)</div>
<div><br>
</div>
<div>What I postulated way back at the beginning of this thread is that the=
 kind of applications that benefit from attribute-based authorization tend =
to be controlled by the web rather than by SIP, and that this is why, altho=
ugh we've been studying this problem
 space for more than a decade, we never standardized an attribute-based aut=
horization protocol for SIP. This entire conferencing example has been abou=
t manipulating conference policy and managing constraints on conference pol=
icy, which may be a problem space
 with unsolved issues (honestly, I'm not much of an expert on the current t=
hinking about conferencing) but I don't think it's the SIP authorization pr=
oblem space.</div>
<div><br>
</div>
<div>The semantics of the INVITE method are &quot;hey, remote resource, I'd=
 like to set up a session with you involving the following media types.&quo=
t; The question of whether or not you accept INVITEs, which is to my mind t=
he SIP authorization problem, is mainly about
 figuring out why you would say yes or no to that question. What you are pr=
oposing here is a conference policy constraint token that lets the IdP pigg=
yback information on the INVITE to be sent to the conference bring. We've s=
truggled here to even identify how
 that information would alter to the decision about whether or not to accep=
t the INVITE. If we need to standardize a protocol mechanism for that funct=
ion, I have a hard time seeing why SIP would be our first choice, and then =
why an INVITE setting up a session
 would be the right choice, versus just doing this with a protocol interact=
ion that has a semantics other than &quot;I'd like to set up a session with=
 you.&quot;</div>
<div><br>
</div>
<div>I'd note that your Google drive example is also a pretty poor fit for =
these semantics. But these are the primary semantics of SIP. If the thing y=
ou want to do has different semantics, there's probably a way to do with a =
different protocol. Albeit, that
 doesn't mean there isn't an authorization problem for SIP worth identifyin=
g and considering.</div>
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<div>Regards,</div>
<div>&nbsp;Rifaat</div>
<div><br>
</div>
<div><br>
</div>
</div>
</div>
</div>
</blockquote>
</span>
</body>
</html>

--_000_D392BAC11A0530jonpetersonneustarbiz_--


From nobody Fri Jun 24 12:13:14 2016
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From: "Roy, Radhika R CIV USARMY RDECOM CERDEC (US)" <radhika.r.roy.civ@mail.mil>
To: "Peterson, Jon" <jon.peterson@neustar.biz>, Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Thread-Topic: [Non-DoD Source] Re: [sipcore] SIP AuthNZ Problem Statement - v3
Thread-Index: AQHRqS0TDyFOI6cZKUWBz4pHTe07DJ+vbNaAgAFjyQCAA2UIgIACvqkAgANXvoCABITogIAH9hmAgAMlg4CAF/VngIAF7xoAgAFfM4CAAEFuAIABZsMAgABovQCADCF4eIAB1wKAgAAE4kA=
Date: Fri, 24 Jun 2016 19:13:01 +0000
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References: <CAGL6ep+8nhiuQs8uBLxmsdwfrxFys4PM8UP5Apbcb7YVK4fmfg@mail.gmail.com> <CAGL6epKhbrdNEgpLFDD+JHFCNcAfST1aCKg3votcVe6z4wVa6A@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B37F97548@ESESSMB209.ericsson.se> <D3562523.18AAA1%jon.peterson@neustar.biz> <CAGL6epJDu1ojsSaPnJqpNoZgCwXvOoFNoEa2ZLNtmRtKMPMyoA@mail.gmail.com> <D35B4CE3.18D9AB%jon.peterson@neustar.biz> <CAGL6epJuByqHi_Fvssyd0V-LE4=_s7HK9DB-9Z3SSVky1Ak4Xw@mail.gmail.com> <D361F10A.18F366%jon.peterson@neustar.biz> <CAGL6epKVaOyVMS_6xmdEXbWsp2UXsbbwjTW2MgM=fJWLE381dg@mail.gmail.com> <D3688271.191B87%jon.peterson@neustar.biz> <CAGL6epKiY_Xi_D+J8QnTfoEyaDU1=jfcbhVFAMY_v3tkQjAhug@mail.gmail.com> <D3842CA2.19BA97%jon.peterson@neustar.biz> <CAGL6epKbdjHP-K9Hjn825HUVTQ4V=TnUYeyrLSU0ETQtmsfs9g@mail.gmail.com> <D385862C.19C10C%jon.peterson@neustar.biz> <CAGL6epLZOhUMQHGGW_Y49gZZAdvFWH07NfO97f=krk1_K9Xd-g@mail.gmail.com> <D386CD3B.19C95F%jon.peterson@neustar.biz> <CAGL6epKj00is8dH7NHdjjuTbh-XbeB7yuALouzR2fzfMBC=R4w@mail.gmail.com> <D38F19C3.19F959%jon.peterson@neustar.biz> <CAGL6epK_DEPr5nYOfGOszLtfT8i8Lso8cZkdXO2+v613ix3eDQ@mail.gmail.com> <D3905929.19FF98%jon.peterson@neustar.biz> <CAGL6epKJQqprV7z3GA-PMeTWvzD_CE7=q-rcv9Tyw-WSmHndyA@mail.gmail.com> <D392BAC1.1A0530%jon.peterson@neustar.biz>
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Subject: Re: [sipcore] [Non-DoD Source] Re: SIP AuthNZ Problem Statement - v3
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Hi, Rifaat:

If we look into SIP XCON RFCs for centralized conferencing in SIP (and even=
 in 3pcc),  the fundamental assumption is this: All conferencing services t=
o all users are centrally controlled by the "focus." It has been done for a=
 very solid technical reason: Distributed real-time conferencing services w=
ith multimedia for multiparty are very difficult to deal with, not only in =
SIP but in all other call control protocols, and security feature is only o=
ne them.

If the above criterion is not met and AuthNZ is put outside the control of =
the focus for real-time multimedia multipoint call, Jon has shown diligentl=
y where it leads to us.

So, I suggest let us analyze all RFCs related of XCON WG and suggest what w=
e can do best here under the umbrella of focus.

BR/Radhika=20

PS: Jon - I got the answer of my earlier email sent to you why focus will n=
ot be able to either accept or reject the call.

-----Original Message-----
From: sipcore [mailto:sipcore-bounces@ietf.org] On Behalf Of Peterson, Jon
Sent: Friday, June 24, 2016 2:27 PM
To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Cc: sipcore@ietf.org
Subject: [Non-DoD Source] Re: [sipcore] SIP AuthNZ Problem Statement - v3



		Moreover, in the ad hoc conference corner case, I see no plausible justif=
ication for why the conference bridge would ever reject Alice's initial INV=
ITE based on the media type it contains. If she's sending her INVITE throug=
h the IdP via SIP, it would never even forward the request, let alone issue=
 a token for it, if it didn't like the media type that Alice was proposing =
- if Alice proposes "video", and the IdP only allows "audio", when the IdP =
receives Alice's INVITE containing "video" it would do the rejecting itself=
 (in SIP or HTTP).=20


	Yes, that is the whole idea; to delegate the authorization part from the c=
onference server; the only thing that the conference will be doing now is e=
nforcing the authorization provided in the token, assuming the token is val=
id and provided by a trusted entity.


... um, I would have thought this totally undermined your argument. If the =
IdP simply prevents the INVITE from ever being sent to the conference bridg=
e when the proposed media type is unacceptable to the IdP, then there's no =
need to carry attribute to the conference bridge at all. If the IdP doesn't=
 like the media type, the conference bridge will never see the INVITE; and =
if the conference bridge sees the INVITE, then the IdP must have liked the =
media type. Attribute adds nothing.

This is a consequence of architecting an administrative domain in a draconi=
an way that lets intermediaries prevent user agents from sending SIP messag=
es where they like, of course. But even if we assume a more open environmen=
t where the user agent can send INVITEs directly to the conference bridge, =
all the conference bridge needs is a signature or similar token that indica=
tes the IdP's approval, -not- any specific attribute.


	=20

		So once again, I don't see how this token would ever lead to the actual I=
NVITE that carried the token ever being rejected by the conference bridge b=
ased on the media types authorized by the token.


	So what? why is this a problem?


It's a problem because you're proposing to take a feature (OAuth) that exis=
ts in HTTP and to add it to SIP.  To make an argument for that, you at leas=
t must establish that SIP needs this feature - and preferably you need to e=
stablish that making this protocol change and performing these operations i=
n SIP is less work (in the grand scheme of implementation and deployment ov=
erall) than saying that if you want to perform operations like this you sho=
uld instead just use HTTP. (Which you can, and for these conference cases, =
you surely should.)

What I postulated way back at the beginning of this thread is that the kind=
 of applications that benefit from attribute-based authorization tend to be=
 controlled by the web rather than by SIP, and that this is why, although w=
e've been studying this problem space for more than a decade, we never stan=
dardized an attribute-based authorization protocol for SIP. This entire con=
ferencing example has been about manipulating conference policy and managin=
g constraints on conference policy, which may be a problem space with unsol=
ved issues (honestly, I'm not much of an expert on the current thinking abo=
ut conferencing) but I don't think it's the SIP authorization problem space=
.

The semantics of the INVITE method are "hey, remote resource, I'd like to s=
et up a session with you involving the following media types." The question=
 of whether or not you accept INVITEs, which is to my mind the SIP authoriz=
ation problem, is mainly about figuring out why you would say yes or no to =
that question. What you are proposing here is a conference policy constrain=
t token that lets the IdP piggyback information on the INVITE to be sent to=
 the conference bring. We've struggled here to even identify how that infor=
mation would alter to the decision about whether or not to accept the INVIT=
E. If we need to standardize a protocol mechanism for that function, I have=
 a hard time seeing why SIP would be our first choice, and then why an INVI=
TE setting up a session would be the right choice, versus just doing this w=
ith a protocol interaction that has a semantics other than "I'd like to set=
 up a session with you."

I'd note that your Google drive example is also a pretty poor fit for these=
 semantics. But these are the primary semantics of SIP. If the thing you wa=
nt to do has different semantics, there's probably a way to do with a diffe=
rent protocol. Albeit, that doesn't mean there isn't an authorization probl=
em for SIP worth identifying and considering.

Jon Peterson
Neustar, Inc.



	Regards,
	 Rifaat




From nobody Fri Jun 24 12:51:00 2016
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Subject: [sipcore] draft-sparks-sipcore-name-addr-guidance-00: comments
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Hi,

The following are some comments concerning
draft-sparks-sipcore-name-addr-guidance-00.

Thanks,
Brett

-------

1) Abstract: "and at least one header field was missed" can potentially be
removed.  The RFC 3261 section 20.31 mandate for Reply-To is the same as
the section 20.20 mandate for From.  However, it was missing from the RFC
3261 section 20 paragraph that this draft is updating.

2) Section 7: Errata 3744 and 3894 are mentioned; however the draft
doesn't explicitly mention RFC 3325.  RFC 3325 potentially should be
mentioned within section 3 (or elsewhere) even if this draft does not need
to update the RFC.


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Subject: [sipcore] The state of draft-ietf-sipcore-dns-dual-stack-07
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You can see the diff between the -06 version of
draft-ietf-sipcore-dns-dual-stack and the current draft of the -07
version using rfcdiff with this URL:

https://www.ietf.org/rfcdiff?url1=draft-ietf-sipcore-dns-dual-stack-06&url2=http://svn.resiprocate.org/rep/ietf-drafts/worley/draft-ietf-sipcore-dns-dual-stack-07.txt

If you have the subversion client, you can see the text diff with

svn diff http://svn.resiprocate.org/rep/ietf-drafts/worley/draft-ietf-sipcore-dns-dual-stack-06.txt http://svn.resiprocate.org/rep/ietf-drafts/worley/draft-ietf-sipcore-dns-dual-stack-07.txt

Dale


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Subject: [sipcore] Happy Eyeballs:  Handoff
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This thread addresses the following requirements.  I consider these to
be part of the Happy Eyeballs problem because they are consequent to
inconsistent access to transit networks.  SIP has these problems,
while HTTP does not, because HTTP does not depend on continuity of
long-lasting connections.  (Thus, these issues are not addressed in
RFC 6555.)

    - Deal with the handoff problem, i.e., when the call (signaling
      and media) are being sent over one interface, and that interface
      becomes unavailable, but another interface has become available,
      the signaling and media should be rerouted via the other
      interface.

    - Similarly, if the external address of a NAT binding changes, the UA
      has, in effect, transitioned from using one interface to another.

      - The primary requirement is that the signaling path is
	reestablished, i.e., the call is not dropped.  It may take several
	seconds for signaling flow to be reestablished.

      - The media flow should be interrupted for only a "short" period of
	time so the user does not assume that the call has been dropped.
	2 seconds seems a reasonable target.

Generally, loss of connectivity can be detected by loss of incoming
RTCP packets.  It looks like the expected RTCP interval is 5 seconds
or longer.  Intermittent loss of RTP due to network congestion is
likely, but we may have to consider detecting loss of RTP as an
indicator of loss of connectivity.  We have to consider both symmetric
loss of connectivity, in which traffic in both directions is lost
simultaneously, and asymmetric loss of connectivity, in which traffic
in one direction is lost while traffic in the other continues.

Restoring RTP (media) connectivity is straightforward once SIP
(signaling) connectivity is restored, by executing a re-INVITE to
renegotiate RTP listening ports, etc.

Restoring signaling connectivity

I can see two ways to restore SIP connectivity:  (1) sending re-INVITE
to perform a target refresh, changing the UA's target URI, and (2)
initiating a new dialog by sending an INVITE-with-Replaces to the
remote target URI in order to replace the dialog with a new dialog.

In either case, the UA should not attempt to modify/replace the dialog
before sending an OPTIONS request and receiving a response from the
new interface to the URI that will be targeted by the new INVITE.
(The round-trip OPTIONS ensures that there is two-way signaling
connectivity to the targeted URI.)  If the UA has more than one
interface that is still working, it probably needs to probe the target
URI using each interface (in parallel), because some URIs may not be
reachable from some interfaces.

Sending a re-INVITE is a good method if the UA knows that the first
URI in the route set can be reached from the UA's new address
(interface).  It seems to me that this will often not be the case,
particularly when handing off between a carrier mobile network and a
private WiFi network.

If the route set of the current dialog cannot be maintained, it is
possible to create an entirely new dialog by directing an
INVITE-with-Replaces to the remote target URI of the dialog.  In a
perfect world, the remote target URI is a GRUU, and the connectivity
of a new INVITE to the GRUU is assured.  Unfortunately there is no
guarantee that will work, either.

The difficulty is that all the UA knows about the dialog is the route
set, and there are no fixed conventions that allow the UA to extract
from the route set a URI that can be targeted by an INVITE/Replaces.
E.g., if the route set is:

    A: UA's target
    B: record route URI 1
    C: record route URI 2
    D: record route URI 3
    E: remote target

It's possible that URI C is the only publicly routable URI, and the
URI for the INVITE/Replaces should be E?Route=C&Route=D.

One possibility is probing each route URI with an OPTIONS request.
That may not be a reliable test if the URI contains an IP address,
especially if the address is in private-use space, as the UA may send
the OPTIONS request to a different server that has the same address.
Though probably if the URI contains a DNS name, then if the OPTIONS
succeeded, it probably reached the same server as the route URI
indicates.

Absent any system for indicating which URIs are publicly routable
(other than the "gr" parameter for GRUUs), we probably have to rely on
the fact that most SIP telephones execute transfers using
INVITE/Replaces requests that assume that the remote target URI that
they see is publicly routable.  As a consequence of this, SIP switches
perform machinations to ensure that the remote target URIs seen by
phones are publicly routable.

Assuming we can assume that remote target URIs are publicly routable,
then we can safely recommend that UAs always use INVITE/Replaces to
restore signaling.

Maintaining one's GRUU

Since we expect a UA to use a GRUU as its target URI so that remote
UAs can target the GRUU to reestablish signaling, a UA must ensure
that its GRUU routes to all the addresses by which it is reachable.
Generally, this means that the UA must update its registration
promptly whenever an interface becomes usable.

However, it looks like there may be some ugly consequences of
maintaining multiple mappings for a UA's GRUU -- how does a request
get routed to the GRUU, serially or parallely?  Can one use
"Request-Disposition: parallel" to force an OPTIONS request to fork
parallely to all of the contacts of a GRUU?  The executing UA does not
need to know which of the contacts of the remote UA were accessible
via the GRUU, but it does need to know quite promptly that some
contact of the remote UA is accessible via the GRUU.

OTOH, when the INVITE/Replaces is processed, we don't want it to be
delayed due to serial forking to contacts that are no longer
accessible, because the timeouts prescribed in SIP are long relative
to the time we want handouts to occur in.  But perhaps
"Request-Disposition: parallel" can be used here, as the first fork of
an INVITE/Replaces to reach a UA will be acted upon and generate a 200
response, and any later arrivals from other forks will receive 481
responses.

Glare

One risk of reestablishing the dialog is that both UAs might attempt
to reestablish the dialog at the same time.  If both UAs attempt to
re-INVITE at the same time, and the invites cross in transit, the
"glare" rules will require each UA to reject the other UA's re-INVITE,
back off, and resend, as described in RFC 3261 section 14.

If one or both UA uses INVITE/Replaces, various conflicts can occur.
It seems to me that the correct way to fix this is to treat the state
of "an INVITE/Replaces to revive the dialog is outstanding" as a
glare-creating condition that is handled the same way as "a re-INVITE
is outstanding".

Charging information

Ideally, if a handoff does not take the call outside the domain of a
single carrier, the carrier should be given enough information to
determine that the new dialog is a logical continuation of the old
dialog, so that it can combine the charging records of the two
dialogs.  In may cases, the carrier can probably determine from the
INVITE/Replaces that the new dialog is related to the old dialog.  But
should there be a rule that requires that the new dialog copy some
charging-related information from the old dialog?

Dale


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Subject: Re: [sipcore] Happy Eyeballs:  Handoff
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> On 27 Jun 2016, at 04:55, Dale R. Worley <worley@ariadne.com> wrote:
> 
> This thread addresses the following requirements.  I consider these to
> be part of the Happy Eyeballs problem because they are consequent to
> inconsistent access to transit networks.  SIP has these problems,
> while HTTP does not, because HTTP does not depend on continuity of
> long-lasting connections.  (Thus, these issues are not addressed in
> RFC 6555.)

Responding about process issues:

I think we should separate our work with Happy Eyeballs from this work.
THis is something missing in Outbound that exist regardless of IPv6.
I think it including it in Happy Eyeballs will make it much harder to 
solve that particular problem - connection setup.

This is connection/flow resume - something that I agree we need to
look at. Switching flows in the middle of a dialog set up with outbound
is needed - as stated above, on multiple flows regardless of address
family. It could be going from mobile data to wifi, from one outbound
flow to a replacement flow (existing or new) or from IPv4 to IPv6,
since IPv4 was deemed to be slower in an ICE background check.

/O

> 
>    - Deal with the handoff problem, i.e., when the call (signaling
>      and media) are being sent over one interface, and that interface
>      becomes unavailable, but another interface has become available,
>      the signaling and media should be rerouted via the other
>      interface.
> 
>    - Similarly, if the external address of a NAT binding changes, the UA
>      has, in effect, transitioned from using one interface to another.
> 
>      - The primary requirement is that the signaling path is
> 	reestablished, i.e., the call is not dropped.  It may take several
> 	seconds for signaling flow to be reestablished.
> 
>      - The media flow should be interrupted for only a "short" period of
> 	time so the user does not assume that the call has been dropped.
> 	2 seconds seems a reasonable target.
> 
> Generally, loss of connectivity can be detected by loss of incoming
> RTCP packets.  It looks like the expected RTCP interval is 5 seconds
> or longer.  Intermittent loss of RTP due to network congestion is
> likely, but we may have to consider detecting loss of RTP as an
> indicator of loss of connectivity.  We have to consider both symmetric
> loss of connectivity, in which traffic in both directions is lost
> simultaneously, and asymmetric loss of connectivity, in which traffic
> in one direction is lost while traffic in the other continues.
> 
> Restoring RTP (media) connectivity is straightforward once SIP
> (signaling) connectivity is restored, by executing a re-INVITE to
> renegotiate RTP listening ports, etc.
> 
> Restoring signaling connectivity
> 
> I can see two ways to restore SIP connectivity:  (1) sending re-INVITE
> to perform a target refresh, changing the UA's target URI, and (2)
> initiating a new dialog by sending an INVITE-with-Replaces to the
> remote target URI in order to replace the dialog with a new dialog.
> 
> In either case, the UA should not attempt to modify/replace the dialog
> before sending an OPTIONS request and receiving a response from the
> new interface to the URI that will be targeted by the new INVITE.
> (The round-trip OPTIONS ensures that there is two-way signaling
> connectivity to the targeted URI.)  If the UA has more than one
> interface that is still working, it probably needs to probe the target
> URI using each interface (in parallel), because some URIs may not be
> reachable from some interfaces.
> 
> Sending a re-INVITE is a good method if the UA knows that the first
> URI in the route set can be reached from the UA's new address
> (interface).  It seems to me that this will often not be the case,
> particularly when handing off between a carrier mobile network and a
> private WiFi network.
> 
> If the route set of the current dialog cannot be maintained, it is
> possible to create an entirely new dialog by directing an
> INVITE-with-Replaces to the remote target URI of the dialog.  In a
> perfect world, the remote target URI is a GRUU, and the connectivity
> of a new INVITE to the GRUU is assured.  Unfortunately there is no
> guarantee that will work, either.
> 
> The difficulty is that all the UA knows about the dialog is the route
> set, and there are no fixed conventions that allow the UA to extract
> from the route set a URI that can be targeted by an INVITE/Replaces.
> E.g., if the route set is:
> 
>    A: UA's target
>    B: record route URI 1
>    C: record route URI 2
>    D: record route URI 3
>    E: remote target
> 
> It's possible that URI C is the only publicly routable URI, and the
> URI for the INVITE/Replaces should be E?Route=C&Route=D.
> 
> One possibility is probing each route URI with an OPTIONS request.
> That may not be a reliable test if the URI contains an IP address,
> especially if the address is in private-use space, as the UA may send
> the OPTIONS request to a different server that has the same address.
> Though probably if the URI contains a DNS name, then if the OPTIONS
> succeeded, it probably reached the same server as the route URI
> indicates.
> 
> Absent any system for indicating which URIs are publicly routable
> (other than the "gr" parameter for GRUUs), we probably have to rely on
> the fact that most SIP telephones execute transfers using
> INVITE/Replaces requests that assume that the remote target URI that
> they see is publicly routable.  As a consequence of this, SIP switches
> perform machinations to ensure that the remote target URIs seen by
> phones are publicly routable.
> 
> Assuming we can assume that remote target URIs are publicly routable,
> then we can safely recommend that UAs always use INVITE/Replaces to
> restore signaling.
> 
> Maintaining one's GRUU
> 
> Since we expect a UA to use a GRUU as its target URI so that remote
> UAs can target the GRUU to reestablish signaling, a UA must ensure
> that its GRUU routes to all the addresses by which it is reachable.
> Generally, this means that the UA must update its registration
> promptly whenever an interface becomes usable.
> 
> However, it looks like there may be some ugly consequences of
> maintaining multiple mappings for a UA's GRUU -- how does a request
> get routed to the GRUU, serially or parallely?  Can one use
> "Request-Disposition: parallel" to force an OPTIONS request to fork
> parallely to all of the contacts of a GRUU?  The executing UA does not
> need to know which of the contacts of the remote UA were accessible
> via the GRUU, but it does need to know quite promptly that some
> contact of the remote UA is accessible via the GRUU.
> 
> OTOH, when the INVITE/Replaces is processed, we don't want it to be
> delayed due to serial forking to contacts that are no longer
> accessible, because the timeouts prescribed in SIP are long relative
> to the time we want handouts to occur in.  But perhaps
> "Request-Disposition: parallel" can be used here, as the first fork of
> an INVITE/Replaces to reach a UA will be acted upon and generate a 200
> response, and any later arrivals from other forks will receive 481
> responses.
> 
> Glare
> 
> One risk of reestablishing the dialog is that both UAs might attempt
> to reestablish the dialog at the same time.  If both UAs attempt to
> re-INVITE at the same time, and the invites cross in transit, the
> "glare" rules will require each UA to reject the other UA's re-INVITE,
> back off, and resend, as described in RFC 3261 section 14.
> 
> If one or both UA uses INVITE/Replaces, various conflicts can occur.
> It seems to me that the correct way to fix this is to treat the state
> of "an INVITE/Replaces to revive the dialog is outstanding" as a
> glare-creating condition that is handled the same way as "a re-INVITE
> is outstanding".
> 
> Charging information
> 
> Ideally, if a handoff does not take the call outside the domain of a
> single carrier, the carrier should be given enough information to
> determine that the new dialog is a logical continuation of the old
> dialog, so that it can combine the charging records of the two
> dialogs.  In may cases, the carrier can probably determine from the
> INVITE/Replaces that the new dialog is related to the old dialog.  But
> should there be a rule that requires that the new dialog copy some
> charging-related information from the old dialog?
> 
> Dale
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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From: worley@ariadne.com (Dale R. Worley)
To: "Olle E. Johansson" <oej@edvina.net>
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Subject: Re: [sipcore] Happy Eyeballs:  Handoff
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"Olle E. Johansson" <oej@edvina.net> writes:
> I think we should separate our work with Happy Eyeballs from this work.
> THis is something missing in Outbound that exist regardless of IPv6.
> I think it including it in Happy Eyeballs will make it much harder to 
> solve that particular problem - connection setup.

It's not even limited to Outbound, either.

But yes, it probably can be profitably separated from Happy Eyeballs
itself.

Dale


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From: worley@ariadne.com (Dale R. Worley)
To: Brett Tate <brett@broadsoft.com>
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Subject: Re: [sipcore] Happy Eyeballs: Targets
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Brett Tate <brett@broadsoft.com> writes:
> The issue with using a smaller T1 value is that it doesn't just impact the
> desired timer.  For instance, it can cause more retries.  Because of this,
> the draft might want to specify allowing the specific smaller timers in a
> way which doesn't rely upon a smaller T1.
>
> If it matters, RFC 3261 section 17.1.1.2 has normative statements
> concerning the topic of lowing T1.

It seems to me that the core concept of that section is "T1 is an
estimate of the round-trip time (RTT)".  And while the average RTT over
the global Internet is nearly 500ms, the RTT to places that expect a lot
of traffic seems to be 100ms or less.  (I expect RTT on carrier-managed
networks to be even less.)  So I don't see cutting the default T1 as
contravening that section.

True, that does increase the number of retries, but that seems to be the
correct change if it's true that if you don't see a response in 100ms,
it is more likely that the request was lost than that it is delayed in
transit.

Do we have any data on what RTT and packet loss is like in real systems?

Dale


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From: Roman Shpount <roman@telurix.com>
Date: Mon, 27 Jun 2016 15:46:57 -0400
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To: "Dale R. Worley" <worley@ariadne.com>
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Subject: Re: [sipcore] Happy Eyeballs: Targets
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On Mon, Jun 27, 2016 at 3:14 PM, Dale R. Worley <worley@ariadne.com> wrote:

> Do we have any data on what RTT and packet loss is like in real systems?
>

This all depends what you mean by the "real systems". For USA hosted PBX
systems working with office based networks, RTT is 70ms (typical delay from
east coast USA to LA datacenter) or less. If service is geo optimized
between two or more data centers, RTT is 40ms or less. Packet loss is
nearly zero. These conditions account for majority of hosted PBX traffic.

If you are dealing with international, RTT is around 500ms or less for most
locations. Brazil to Singapore is around 420ms RTT, for instance. Western
Europe to LA is around 150ms RTT. There are, of cause, locations where RTT
is much higher, especially when satellite links are involved.

Packet loss heavily depends on the last mile link utilization. There are
still links which are 128K which are used for VoIP. If someone starts
uploading things on this link, you start seeing packet loss or 2-3 sec
packet delays. Most of the time, form locations with broadband connections,
packet loss is close to zero.

If you start dealing with mobile, things are a lot less predictable. Packet
loss can be in 10-20% range. RTT delays can be up to a second. Some other
locations, such as hotels, especially internationally, are often even worse
then mobile due to random traffic blocking or deliberate attempts to block
VoIP.

Please let me know if you need any more data.
_____________
Roman Shpount

--94eb2c11424681f7ab053647c81b
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure" data-smartmail=3D"gmail_signature">On Mon, Jun 27, 2016 at 3:14 PM, Da=
le R. Worley <span dir=3D"ltr">&lt;<a href=3D"mailto:worley@ariadne.com" ta=
rget=3D"_blank">worley@ariadne.com</a>&gt;</span> wrote:<br></div></div><di=
v class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin:0=
px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">D=
o we have any data on what RTT and packet loss is like in real systems?<br>=
</blockquote><div><br></div><div>This all depends what you mean by the &quo=
t;real systems&quot;. For USA hosted PBX systems working with office based =
networks, RTT is 70ms (typical delay from east coast USA to LA datacenter) =
or less. If service is geo optimized between two or more data centers, RTT =
is 40ms or less. Packet loss is nearly zero. These conditions account for m=
ajority of hosted PBX traffic.</div><div><br></div><div>If you are dealing =
with international, RTT is around 500ms or less for most locations. Brazil =
to Singapore is around 420ms RTT, for instance. Western Europe to LA is aro=
und 150ms RTT. There are, of cause, locations where RTT is much higher, esp=
ecially when satellite links are involved.</div><div><br></div><div>Packet =
loss heavily depends on the last mile link utilization. There are still lin=
ks which are 128K which are used for VoIP. If someone starts uploading thin=
gs on this link, you start seeing packet loss or 2-3 sec packet delays. Mos=
t of the time, form locations with broadband connections, packet loss is cl=
ose to zero.</div><div><br></div><div>If you start dealing with mobile, thi=
ngs are a lot less predictable. Packet loss can be in 10-20% range. RTT del=
ays can be up to a second. Some other locations, such as hotels, especially=
 internationally, are often even worse then mobile due to random traffic bl=
ocking or deliberate attempts to block VoIP.</div><div><br></div><div>Pleas=
e let me know if you need any more data.</div><div><div class=3D"gmail_sign=
ature" data-smartmail=3D"gmail_signature">_____________<br>Roman Shpount</d=
iv></div><div>=C2=A0</div></div></div></div>

--94eb2c11424681f7ab053647c81b--


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In-Reply-To: <D392BAC1.1A0530%jon.peterson@neustar.biz>
References: <CAGL6ep+8nhiuQs8uBLxmsdwfrxFys4PM8UP5Apbcb7YVK4fmfg@mail.gmail.com> <CAGL6epKhbrdNEgpLFDD+JHFCNcAfST1aCKg3votcVe6z4wVa6A@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B37F97548@ESESSMB209.ericsson.se> <D3562523.18AAA1%jon.peterson@neustar.biz> <CAGL6epJDu1ojsSaPnJqpNoZgCwXvOoFNoEa2ZLNtmRtKMPMyoA@mail.gmail.com> <D35B4CE3.18D9AB%jon.peterson@neustar.biz> <CAGL6epJuByqHi_Fvssyd0V-LE4=_s7HK9DB-9Z3SSVky1Ak4Xw@mail.gmail.com> <D361F10A.18F366%jon.peterson@neustar.biz> <CAGL6epKVaOyVMS_6xmdEXbWsp2UXsbbwjTW2MgM=fJWLE381dg@mail.gmail.com> <D3688271.191B87%jon.peterson@neustar.biz> <CAGL6epKiY_Xi_D+J8QnTfoEyaDU1=jfcbhVFAMY_v3tkQjAhug@mail.gmail.com> <D3842CA2.19BA97%jon.peterson@neustar.biz> <CAGL6epKbdjHP-K9Hjn825HUVTQ4V=TnUYeyrLSU0ETQtmsfs9g@mail.gmail.com> <D385862C.19C10C%jon.peterson@neustar.biz> <CAGL6epLZOhUMQHGGW_Y49gZZAdvFWH07NfO97f=krk1_K9Xd-g@mail.gmail.com> <D386CD3B.19C95F%jon.peterson@neustar.biz> <CAGL6epKj00is8dH7NHdjjuTbh-XbeB7yuALouzR2fzfMBC=R4w@mail.gmail.com> <D38F19C3.19F959%jon.peterson@neustar.biz> <CAGL6epK_DEPr5nYOfGOszLtfT8i8Lso8cZkdXO2+v613ix3eDQ@mail.gmail.com> <D3905929.19FF98%jon.peterson@neustar.biz> <CAGL6epKJQqprV7z3GA-PMeTWvzD_CE7=q-rcv9Tyw-WSmHndyA@mail.gmail.com> <D392BAC1.1A0530%jon.peterson@neustar.biz>
From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Date: Tue, 28 Jun 2016 05:32:21 -0400
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To: "Peterson, Jon" <jon.peterson@neustar.biz>
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Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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On Fri, Jun 24, 2016 at 2:27 PM, Peterson, Jon <jon.peterson@neustar.biz>
wrote:

>
>>
>> Moreover, in the ad hoc conference corner case, I see no plausible
>> justification for why the conference bridge would ever reject Alice's
>> initial INVITE based on the media type it contains. If she's sending her
>> INVITE through the IdP via SIP, it would never even forward the request,
>> let alone issue a token for it, if it didn't like the media type that Alice
>> was proposing - if Alice proposes "video", and the IdP only allows "audio",
>> when the IdP receives Alice's INVITE containing "video" it would do the
>> rejecting itself (in SIP or HTTP).
>>
>
> Yes, that is the whole idea; to delegate the authorization part from the
> conference server; the only thing that the conference will be doing now is
> enforcing the authorization provided in the token, assuming the token is
> valid and provided by a trusted entity.
>
>
> ... um, I would have thought this totally undermined your argument. If the
> IdP simply prevents the INVITE from ever being sent to the conference
> bridge when the proposed media type is unacceptable to the IdP, then
> there's no need to carry attribute to the conference bridge at all. If the
> IdP doesn't like the media type, the conference bridge will never see the
> INVITE; and if the conference bridge sees the INVITE, then the IdP must
> have liked the media type. Attribute adds nothing.
>
> This is a consequence of architecting an administrative domain in a
> draconian way that lets intermediaries prevent user agents from sending SIP
> messages where they like, of course.
>

I am not sure what you mean by this. The IdP cannot prevent anybody from
sending any request to any server.



> But even if we assume a more open environment where the user agent can
> send INVITEs directly to the conference bridge, all the conference bridge
> needs is a signature or similar token that indicates the IdP's approval,
> -not- any specific attribute.
>

Of course attributes are needed.
How would otherwise the conference server know which type of conference to
start (audio, video, etc)? what is the limitation on the number of
participant? etc...



>
>
>
>> So once again, I don't see how this token would ever lead to the actual
>> INVITE that carried the token ever being rejected by the conference bridge
>> based on the media types authorized by the token.
>>
>
> So what? why is this a problem?
>
>
> It's a problem because you're proposing to take a feature (OAuth) that
> exists in HTTP and to add it to SIP.
>

I would not call OAuth a "feature". It is an authorization framework.



> To make an argument for that, you at least must establish that SIP needs
> this feature - and preferably you need to establish that making this
> protocol change and performing these operations in SIP is less work (in the
> grand scheme of implementation and deployment overall) than saying that if
> you want to perform operations like this you should instead just use HTTP.
> (Which you can, and for these conference cases, you surely should.)
>
> What I postulated way back at the beginning of this thread is that the
> kind of applications that benefit from attribute-based authorization tend
> to be controlled by the web rather than by SIP, and that this is why,
> although we've been studying this problem space for more than a decade, we
> never standardized an attribute-based authorization protocol for SIP. This
> entire conferencing example has been about manipulating conference policy
> and managing constraints on conference policy, which may be a problem space
> with unsolved issues (honestly, I'm not much of an expert on the current
> thinking about conferencing) but I don't think it's the SIP authorization
> problem space.
>
> The semantics of the INVITE method are "hey, remote resource, I'd like to
> set up a session with you involving the following media types." The
> question of whether or not you accept INVITEs, which is to my mind the SIP
> authorization problem,
>

This is the main difference between how you and I see this.
I think that the authorization problem is much bigger than just authorizing
a specific SIP method. The authorization should be dealing with *access to
services* provided by SIP servers regardless of the SIP method being used.
Much the same way that HTTP authorizes *access to resources*, regardless of
the HTTP method being used. You do not need a specific token for HTTP GET
vs HTTP POST, because the method does not matter; what matters is the
resource that the method is trying to access.

Regards,
 Rifaat




> is mainly about figuring out why you would say yes or no to that question.
> What you are proposing here is a conference policy constraint token that
> lets the IdP piggyback information on the INVITE to be sent to the
> conference bring. We've struggled here to even identify how that
> information would alter to the decision about whether or not to accept the
> INVITE. If we need to standardize a protocol mechanism for that function, I
> have a hard time seeing why SIP would be our first choice, and then why an
> INVITE setting up a session would be the right choice, versus just doing
> this with a protocol interaction that has a semantics other than "I'd like
> to set up a session with you."
>
> I'd note that your Google drive example is also a pretty poor fit for
> these semantics. But these are the primary semantics of SIP. If the thing
> you want to do has different semantics, there's probably a way to do with a
> different protocol. Albeit, that doesn't mean there isn't an authorization
> problem for SIP worth identifying and considering.
>
> Jon Peterson
> Neustar, Inc.
>
>
> Regards,
>  Rifaat
>
>
>

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

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Fri, Jun 24, 2016 at 2:27 PM, Peterson, Jon <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:jon.peterson@neustar.biz" target=3D"_blank">jon.peterson@ne=
ustar.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-style:solid;=
border-left-color:rgb(204,204,204);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>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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>
<br>
</div>
<div>Moreover, in the ad hoc conference corner case, I see no plausible jus=
tification for why the conference bridge would ever reject Alice&#39;s init=
ial INVITE based on the media type it contains. If she&#39;s sending her IN=
VITE through the IdP via SIP, it would never
 even forward the request, let alone issue a token for it, if it didn&#39;t=
 like the media type that Alice was proposing - if Alice proposes &quot;vid=
eo&quot;, and the IdP only allows &quot;audio&quot;, when the IdP receives =
Alice&#39;s INVITE containing &quot;video&quot; it would do the rejecting
 itself (in SIP or HTTP). </div>
</div>
</blockquote>
<div><br>
</div>
<div>
<div>Yes, that is the whole idea; to delegate the authorization part from t=
he conference server; the only thing that the conference will be doing now =
is enforcing the authorization provided in the token, assuming the token is=
 valid and provided by a trusted
 entity.</div>
</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span><div>... um, I would have thought this totally undermined your argum=
ent. If the IdP simply prevents the INVITE from ever being sent to the conf=
erence bridge when the proposed media type is unacceptable to the IdP, then=
 there&#39;s no need to carry attribute to
 the conference bridge at all. If the IdP doesn&#39;t like the media type, =
the conference bridge will never see the INVITE; and if the conference brid=
ge sees the INVITE, then the IdP must have liked the media type. Attribute =
adds nothing.</div>
<div><br>
</div>
<div>This is a consequence of architecting an administrative domain in a dr=
aconian way that lets intermediaries prevent user agents from sending SIP m=
essages where they like, of course. </div></div></blockquote><div><br></div=
><div>I am not sure what you mean by this. The IdP cannot prevent anybody f=
rom sending any request to any server.</div><div><br></div><div>=C2=A0</div=
><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border=
-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);=
padding-left:1ex"><div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-=
size:14px;font-family:Calibri,sans-serif"><div>But even if we assume a more=
 open environment where the user agent can
 send INVITEs directly to the conference bridge, all the conference bridge =
needs is a signature or similar token that indicates the IdP&#39;s approval=
, -not- any specific attribute.</div></div></blockquote><div><br></div><div=
>Of course attributes are needed.=C2=A0</div><div>How would otherwise the c=
onference server know which type of conference to start (audio, video, etc)=
? what is the limitation on the number of participant? etc...</div><div><br=
></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0=
px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-=
color:rgb(204,204,204);padding-left:1ex"><div style=3D"word-wrap:break-word=
;color:rgb(0,0,0);font-size:14px;font-family:Calibri,sans-serif"><span>
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>=C2=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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>So once again, I don&#39;t see how this token would ever lead to the a=
ctual INVITE that carried the token ever being rejected by the conference b=
ridge based on the media types authorized by the token.</div>
</div>
</blockquote>
<div><br>
</div>
<div>So what? why is this a problem?</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
</span><div>It&#39;s a problem because you&#39;re proposing to take a featu=
re (OAuth) that exists in HTTP and to add it to SIP. =C2=A0</div></div></bl=
ockquote><div><br></div><div>I would not call OAuth a &quot;feature&quot;. =
It is an authorization framework.</div><div><br></div><div>=C2=A0</div><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left=
-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);paddi=
ng-left:1ex"><div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:=
14px;font-family:Calibri,sans-serif"><div>To make an argument for that, you=
 at least must establish that SIP needs this feature - and preferably you n=
eed to establish that making this protocol
 change and performing these operations in SIP is less work (in the grand s=
cheme of implementation and deployment overall) than saying that if you wan=
t to perform operations like this you should instead just use HTTP. (Which =
you can, and for these conference
 cases, you surely should.)</div>
<div><br>
</div>
<div>What I postulated way back at the beginning of this thread is that the=
 kind of applications that benefit from attribute-based authorization tend =
to be controlled by the web rather than by SIP, and that this is why, altho=
ugh we&#39;ve been studying this problem
 space for more than a decade, we never standardized an attribute-based aut=
horization protocol for SIP. This entire conferencing example has been abou=
t manipulating conference policy and managing constraints on conference pol=
icy, which may be a problem space
 with unsolved issues (honestly, I&#39;m not much of an expert on the curre=
nt thinking about conferencing) but I don&#39;t think it&#39;s the SIP auth=
orization problem space.</div>
<div><br>
</div>
<div>The semantics of the INVITE method are &quot;hey, remote resource, I&#=
39;d like to set up a session with you involving the following media types.=
&quot; The question of whether or not you accept INVITEs, which is to my mi=
nd the SIP authorization problem,</div></div></blockquote><div><br></div><d=
iv>This is the main difference between how you and I see this.</div><div>I =
think that the authorization problem is much bigger than just authorizing a=
 specific SIP method. The authorization should be dealing with <b>access to=
 services</b> provided by SIP servers regardless of the SIP method being us=
ed.</div><div>Much the same way that HTTP authorizes <b>access to resources=
</b>, regardless of the HTTP method being used. You do not need a specific =
token for HTTP GET vs HTTP POST, because the method does not matter; what m=
atters is the resource that the method is trying to access.</div><div><br><=
/div><div>Regards,</div><div>=C2=A0Rifaat</div><div><br></div><div><br></di=
v><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0p=
x 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-color=
:rgb(204,204,204);padding-left:1ex"><div style=3D"word-wrap:break-word;colo=
r:rgb(0,0,0);font-size:14px;font-family:Calibri,sans-serif"><div> is mainly=
 about
 figuring out why you would say yes or no to that question. What you are pr=
oposing here is a conference policy constraint token that lets the IdP pigg=
yback information on the INVITE to be sent to the conference bring. We&#39;=
ve struggled here to even identify how
 that information would alter to the decision about whether or not to accep=
t the INVITE. If we need to standardize a protocol mechanism for that funct=
ion, I have a hard time seeing why SIP would be our first choice, and then =
why an INVITE setting up a session
 would be the right choice, versus just doing this with a protocol interact=
ion that has a semantics other than &quot;I&#39;d like to set up a session =
with you.&quot;</div>
<div><br>
</div>
<div>I&#39;d note that your Google drive example is also a pretty poor fit =
for these semantics. But these are the primary semantics of SIP. If the thi=
ng you want to do has different semantics, there&#39;s probably a way to do=
 with a different protocol. Albeit, that
 doesn&#39;t mean there isn&#39;t an authorization problem for SIP worth id=
entifying and considering.</div><span>
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
<div><br>
</div>
<span>
<blockquote style=3D"BORDER-LEFT:#b5c4df 5 solid;PADDING:0 0 0 5;MARGIN:0 0=
 0 5">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<div>Regards,</div>
<div>=C2=A0Rifaat</div>
<div><br>
</div>
<div><br>
</div>
</div>
</div>
</div>
</blockquote>
</span>
</span></div>

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

--001a114161185f21c605365350a8--


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To: "Dale R. Worley" <worley@ariadne.com>, sipcore@ietf.org
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Subject: Re: [sipcore] Happy Eyeballs: Targets
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Hi,

Thanks for the response; reply is inline.

> > The issue with using a smaller T1 value is that it doesn't
> > just impact the desired timer.  For instance, it can cause
> > more retries.  Because of this, the draft might want to
> > specify allowing the specific smaller timers in a way which
> > doesn't rely upon a smaller T1.
> >
> > If it matters, RFC 3261 section 17.1.1.2 has normative
> > statements concerning the topic of lowing T1.
>
> It seems to me that the core concept of that section is
> "T1 is an estimate of the round-trip time (RTT)".
> And while the average RTT over the global Internet is nearly
> 500ms, the RTT to places that expect a lot of traffic
> seems to be 100ms or less.  (I expect RTT on carrier-managed
> networks to be even less.)  So I don't see cutting the
> default T1 as contravening that section.

Although this draft can update it if needed, RFC 3261 section 17.1.1.2
paragraph four was the text of potential concern since it looks like an
attempt to limit where T1 can be lowered.

"Elements MAY (though it
is NOT RECOMMENDED) use smaller values of T1 within closed, private
networks that do not permit general Internet connection.  T1 MAY be
chosen larger, and this is RECOMMENDED if it is known in advance
(such as on high latency access links) that the RTT is larger."


> True, that does increase the number of retries, but that seems
> to be the correct change if it's true that if you don't see a
> response in 100ms, it is more likely that the request was lost
> than that it is delayed in transit.

It seems like the extra retries would be an unnecessary burden upon the
next hop.  However, I guess that it depends upon if more concerned with 1)
dropped packets or 2) the retries increasing the frequency of devices
becoming overloaded.

Unless this draft updates the RFC 4320 behavior, there can be many retries
before the next hop can return 100 for non-INVITE requests.

As mentioned, the smaller T1 value would not just impact the desired
timers.  It would impact other letter timers within RFC 3261 and RFC 6026
such as Timer L, Timer M, Timer H, Timer J, and as mentioned Timer A,
Timer E, and Timer G.  Should these timers really be reduced?  For
instance, is it acceptable for UAS to use T1 of 100ms when the UAC and
intermediaries are using T1 of 500ms?


From nobody Tue Jun 28 12:04:16 2016
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From: worley@ariadne.com (Dale R. Worley)
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Subject: [sipcore] How to construct a finite state machine to select an alerting signal
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I've worked out how to construct a finite state machine to process the
'alert' URNs in an Alert-Info header to select one of the alerting
signals of a UA.  Once you fix the set of signals and their meanings
(the URNs that the signal indicates), there's an algorithm that
constructs the FSM, and it ensures that FSM satisfies the rules in RFC
7462.

I've written a draft to describe the process in detail.  The draft also
points to an implementation of the algorithm in Python, so people can
try out this method and see how it works for them.

I'd like to hear any feedback you have.

Dale


A new version of I-D, draft-worley-alert-info-fsm-02.txt
has been successfully submitted by Dale R. Worley and posted to the
IETF repository.

Name:		draft-worley-alert-info-fsm
Revision:	02
Title:		A Simpler Method for Processing Alert-Info URNs
Document date:	2016-06-05
Group:		Individual Submission
Pages:		36
URL:            https://www.ietf.org/internet-drafts/draft-worley-alert-info-fsm-02.txt
Status:         https://datatracker.ietf.org/doc/draft-worley-alert-info-fsm/
Htmlized:       https://tools.ietf.org/html/draft-worley-alert-info-fsm-02
Diff:           https://www.ietf.org/rfcdiff?url2=draft-worley-alert-info-fsm-02

Abstract:
   The "alert" namespace of uniform resource names (URNs) can be used in
   the Alert-Info header field of Session Initiation Protocol (SIP)
   requests and responses to inform a VoIP telephone (user agent) of the
   characteristics of the call that the user agent has originated or
   terminated.  Based on the URNs in the Alert-Info header field, the
   user agent must select an the best available signal to present to its
   user to indicate the characteristics of the call.  This document
   describes a method by which a user agent's designer can, based on the
   user agent's signals and their meanings, constructing a finite state
   machine (FSM) to process the URNs to select a signal in a way that
   obeys the restrictions given in the definition of the "alert" URN
   namespace.  In many situations, the resulting FSM is simpler and
   faster than the previously described selection algorithm.


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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Date: Tue, 28 Jun 2016 16:55:07 -0400
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Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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On 6/28/16 5:32 AM, Rifaat Shekh-Yusef wrote:

>     The semantics of the INVITE method are "hey, remote resource, I'd
>     like to set up a session with you involving the following media
>     types." The question of whether or not you accept INVITEs, which is
>     to my mind the SIP authorization problem,
>
> This is the main difference between how you and I see this.
> I think that the authorization problem is much bigger than just
> authorizing a specific SIP method. The authorization should be dealing
> with *access to services* provided by SIP servers regardless of the SIP
> method being used.
> Much the same way that HTTP authorizes *access to resources*, regardless
> of the HTTP method being used. You do not need a specific token for HTTP
> GET vs HTTP POST, because the method does not matter; what matters is
> the resource that the method is trying to access.

If you want to tell a server something about what you want it to do, you 
can use caller preferences. At least then you are being explicit about it.

That seems more straightforward than embedding similar information into 
an authorization token.

	Thanks,
	Paul


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From: worley@ariadne.com (Dale R. Worley)
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Subject: [sipcore] Happy Eyeballs:  Outbounde
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This thread is to explore "outbound" issues.  As I have them, the
requirements are:

    - Clients behind client-side NATs must be fully supported.

    - The solution should be as compatible with SIP Outbound as practicable.

    The difficult part of this is "the solution for simple UDP (i.e., not
    using SIP Outbound)".  There is a strong perception that in systems
    where an edge proxy services 10^5 or 10^6 UAs that only
    UDP-without-Outbound has a low enough per-flow overhead to be
    workable.  The important requirements for simple UDP seem to be:

    - The per-UA state in the edge proxy must be no larger than about what
      is kept for a registration.

    - The update rate of the per-UA state in the edge proxy must be no
      larger than about the update rate for a registration.

    It is perceived that TCP, TLS, and even UDP-with-Outbound do not meet
    these requirements.  A substantial fraction of the UAs tested in the
    latest SIPit do not support TCP, TLS, or Outbound, showing that there
    is substantial market presence of simple-UDP-only UAs.

    It is possible that a "Simple Outbound" can be defined that provides
    the needed part of the functionality of Outbound at a sufficiently low
    overhead and complexity that it meets these requirements.  If so, it
    is desirable that it be conceptually and operationally
    upward-compatible with Outbound.

It's clear to me that the problem is *solvable*, because existing SIP
systems do handle the client-side NAT problem.  E.g., the open-source
sipX system has full client-side NAT support.  That scheme doesn't
require SIP Outbound support in the client at all.  NAT support is
triggered by the client's requests arriving from an address that is
different than what is specified in the request.  Support is
implemented by manipulating the client's behavior, rewriting
requests/responses to substitute IP addresses, and providing
(essentially) a TURN server to relay media.

As far as I can remember, sipX's NAT support is recorded and
implemented in the standard registration/redirect database.  However,
NAT support does depend on forcing the client to re-register
frequently enough to be assured that the NAT mapping is not released.
Since processing re-registrations is by far the bulk of the signaling
traffic even without NAT support, this is not a trivial change.

My expectation is that almost all commercial SIP systems have NAT
support of this sort.

One difference between this sort of NAT support and Outbound is that
NAT support is done only at the registrar/proxy; if there is a
separate edge proxy, it only passes UDP messages and can easily be
stateless.  This might be a significant factor in very large
deployments.

Perhaps a significant problem with Outbound is that it has to be
implemented in both the phone and the switch, leading to a network
effect problem.

At this point, it seems to me that we need to get a better
understanding of what people are doing in the market to deal with NATs
and find out why they don't use Outbound.  (Since Outbound is the
standard method, I would think it has a strategic advantage in the
technological competition.)

Dale


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From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Date: Thu, 30 Jun 2016 10:29:13 -0400
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To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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An access token is explicit too; it carries authorization to access some
resource/service.

I am not sure what you have in mind.
Are you suggesting that after the UA/Proxy gets the token from the IdP,
that the UA/Proxy would then transform the authorizations given in the
token to a SIP caller preference format?

Regards,
 Rifaat




On Tue, Jun 28, 2016 at 4:55 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> On 6/28/16 5:32 AM, Rifaat Shekh-Yusef wrote:
>
>     The semantics of the INVITE method are "hey, remote resource, I'd
>>     like to set up a session with you involving the following media
>>     types." The question of whether or not you accept INVITEs, which is
>>     to my mind the SIP authorization problem,
>>
>> This is the main difference between how you and I see this.
>> I think that the authorization problem is much bigger than just
>> authorizing a specific SIP method. The authorization should be dealing
>> with *access to services* provided by SIP servers regardless of the SIP
>> method being used.
>> Much the same way that HTTP authorizes *access to resources*, regardless
>> of the HTTP method being used. You do not need a specific token for HTTP
>> GET vs HTTP POST, because the method does not matter; what matters is
>> the resource that the method is trying to access.
>>
>
> If you want to tell a server something about what you want it to do, you
> can use caller preferences. At least then you are being explicit about it.
>
> That seems more straightforward than embedding similar information into an
> authorization token.
>
>         Thanks,
>         Paul
>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

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<div dir=3D"ltr">An access token is explicit too; it carries authorization =
to access some resource/service.<div><br></div><div>I am not sure what you =
have in mind.=C2=A0</div><div>Are you suggesting that after the UA/Proxy ge=
ts the token from the IdP, that the UA/Proxy would then transform the autho=
rizations given in the token to a SIP caller preference format?</div><div><=
br></div><div>Regards,</div><div>=C2=A0Rifaat</div><div><br></div><div><br>=
<div><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Tue, =
Jun 28, 2016 at 4:55 PM, Paul Kyzivat <span dir=3D"ltr">&lt;<a href=3D"mail=
to:pkyzivat@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit.edu</a>&gt;</=
span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8e=
x;border-left:1px #ccc solid;padding-left:1ex"><span>On 6/28/16 5:32 AM, Ri=
faat Shekh-Yusef wrote:<br>
<br>
</span><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-=
left:1px #ccc solid;padding-left:1ex"><span>
=C2=A0 =C2=A0 The semantics of the INVITE method are &quot;hey, remote reso=
urce, I&#39;d<br>
=C2=A0 =C2=A0 like to set up a session with you involving the following med=
ia<br>
=C2=A0 =C2=A0 types.&quot; The question of whether or not you accept INVITE=
s, which is<br>
=C2=A0 =C2=A0 to my mind the SIP authorization problem,<br>
<br>
This is the main difference between how you and I see this.<br>
I think that the authorization problem is much bigger than just<br>
authorizing a specific SIP method. The authorization should be dealing<br><=
/span>
with *access to services* provided by SIP servers regardless of the SIP<br>
method being used.<br>
Much the same way that HTTP authorizes *access to resources*, regardless<sp=
an><br>
of the HTTP method being used. You do not need a specific token for HTTP<br=
>
GET vs HTTP POST, because the method does not matter; what matters is<br>
the resource that the method is trying to access.<br>
</span></blockquote>
<br>
If you want to tell a server something about what you want it to do, you ca=
n use caller preferences. At least then you are being explicit about it.<br=
>
<br>
That seems more straightforward than embedding similar information into an =
authorization token.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Thanks,<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Paul<div><div><br>
<br>
_______________________________________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a><=
br>
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</div></div></blockquote></div><br></div></div></div></div>

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To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
References: <CAGL6ep+8nhiuQs8uBLxmsdwfrxFys4PM8UP5Apbcb7YVK4fmfg@mail.gmail.com> <D3688271.191B87%jon.peterson@neustar.biz> <CAGL6epKiY_Xi_D+J8QnTfoEyaDU1=jfcbhVFAMY_v3tkQjAhug@mail.gmail.com> <D3842CA2.19BA97%jon.peterson@neustar.biz> <CAGL6epKbdjHP-K9Hjn825HUVTQ4V=TnUYeyrLSU0ETQtmsfs9g@mail.gmail.com> <D385862C.19C10C%jon.peterson@neustar.biz> <CAGL6epLZOhUMQHGGW_Y49gZZAdvFWH07NfO97f=krk1_K9Xd-g@mail.gmail.com> <D386CD3B.19C95F%jon.peterson@neustar.biz> <CAGL6epKj00is8dH7NHdjjuTbh-XbeB7yuALouzR2fzfMBC=R4w@mail.gmail.com> <D38F19C3.19F959%jon.peterson@neustar.biz> <CAGL6epK_DEPr5nYOfGOszLtfT8i8Lso8cZkdXO2+v613ix3eDQ@mail.gmail.com> <D3905929.19FF98%jon.peterson@neustar.biz> <CAGL6epKJQqprV7z3GA-PMeTWvzD_CE7=q-rcv9Tyw-WSmHndyA@mail.gmail.com> <D392BAC1.1A0530%jon.peterson@neustar.biz> <CAGL6ep++QG3uKt0Rtib_eXQVqJQS1Amr5njz1os3WT6xDy5T-w@mail.gmail.com> <b8f18bd8-931c-4730-02b0-f36c519d2943@alum.mit.edu> <CAGL6epJYLKNSTa8C8xka47mFot2WES_rEyw_6Shz6RsSkhDX_w@mail.gmail.com>
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
Message-ID: <1a0ef7d8-4fb7-166d-4e9f-25e200eee1e2@alum.mit.edu>
Date: Thu, 30 Jun 2016 10:43:20 -0400
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Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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On 6/30/16 10:29 AM, Rifaat Shekh-Yusef wrote:
> An access token is explicit too; it carries authorization to access some
> resource/service.
>
> I am not sure what you have in mind.
> Are you suggesting that after the UA/Proxy gets the token from the IdP,
> that the UA/Proxy would then transform the authorizations given in the
> token to a SIP caller preference format?

It sounds like you are imagining a situation where the user gets a token 
that authorizes some specific service(s) without the user knowing what 
services those are. And then the user makes a request with the token and 
expects the server to figure out what services to supply.

That seems backwards to me. I would expect that the user would figure 
out what services he wants, then get some credentials that are 
sufficient to authorize the service, and then request the service and 
supply the credentials to cover it.

It may be that the credentials he has are broad enough to cover lots of 
services. He doesn't want them all, he wants what he wants at the time 
of invocation.

So callerprefs provides a mechanism to express what features you want 
from a callee. To that you then need to supply credentials, in this case 
in the form of an authorization token.

	Thanks,
	Paul

> Regards,
>  Rifaat
>
>
>
>
> On Tue, Jun 28, 2016 at 4:55 PM, Paul Kyzivat <pkyzivat@alum.mit.edu
> <mailto:pkyzivat@alum.mit.edu>> wrote:
>
>     On 6/28/16 5:32 AM, Rifaat Shekh-Yusef wrote:
>
>             The semantics of the INVITE method are "hey, remote
>         resource, I'd
>             like to set up a session with you involving the following media
>             types." The question of whether or not you accept INVITEs,
>         which is
>             to my mind the SIP authorization problem,
>
>         This is the main difference between how you and I see this.
>         I think that the authorization problem is much bigger than just
>         authorizing a specific SIP method. The authorization should be
>         dealing
>         with *access to services* provided by SIP servers regardless of
>         the SIP
>         method being used.
>         Much the same way that HTTP authorizes *access to resources*,
>         regardless
>         of the HTTP method being used. You do not need a specific token
>         for HTTP
>         GET vs HTTP POST, because the method does not matter; what
>         matters is
>         the resource that the method is trying to access.
>
>
>     If you want to tell a server something about what you want it to do,
>     you can use caller preferences. At least then you are being explicit
>     about it.
>
>     That seems more straightforward than embedding similar information
>     into an authorization token.
>
>             Thanks,
>             Paul
>
>
>     _______________________________________________
>     sipcore mailing list
>     sipcore@ietf.org <mailto:sipcore@ietf.org>
>     https://www.ietf.org/mailman/listinfo/sipcore
>
>


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References: <CAGL6ep+8nhiuQs8uBLxmsdwfrxFys4PM8UP5Apbcb7YVK4fmfg@mail.gmail.com> <D3688271.191B87%jon.peterson@neustar.biz> <CAGL6epKiY_Xi_D+J8QnTfoEyaDU1=jfcbhVFAMY_v3tkQjAhug@mail.gmail.com> <D3842CA2.19BA97%jon.peterson@neustar.biz> <CAGL6epKbdjHP-K9Hjn825HUVTQ4V=TnUYeyrLSU0ETQtmsfs9g@mail.gmail.com> <D385862C.19C10C%jon.peterson@neustar.biz> <CAGL6epLZOhUMQHGGW_Y49gZZAdvFWH07NfO97f=krk1_K9Xd-g@mail.gmail.com> <D386CD3B.19C95F%jon.peterson@neustar.biz> <CAGL6epKj00is8dH7NHdjjuTbh-XbeB7yuALouzR2fzfMBC=R4w@mail.gmail.com> <D38F19C3.19F959%jon.peterson@neustar.biz> <CAGL6epK_DEPr5nYOfGOszLtfT8i8Lso8cZkdXO2+v613ix3eDQ@mail.gmail.com> <D3905929.19FF98%jon.peterson@neustar.biz> <CAGL6epKJQqprV7z3GA-PMeTWvzD_CE7=q-rcv9Tyw-WSmHndyA@mail.gmail.com> <D392BAC1.1A0530%jon.peterson@neustar.biz> <CAGL6ep++QG3uKt0Rtib_eXQVqJQS1Amr5njz1os3WT6xDy5T-w@mail.gmail.com> <b8f18bd8-931c-4730-02b0-f36c519d2943@alum.mit.edu> <CAGL6epJYLKNSTa8C8xka47mFot2WES_rEyw_6Shz6RsSkhDX_w@mail.gmail.com> <1a0ef7d8-4fb7-166d-4e9f-25e200eee1e2@alum.mit.edu>
From: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Date: Thu, 30 Jun 2016 13:02:29 -0400
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To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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--001a113ad63cf4fde2053681d5c2
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On Thu, Jun 30, 2016 at 10:43 AM, Paul Kyzivat <pkyzivat@alum.mit.edu>
wrote:

> On 6/30/16 10:29 AM, Rifaat Shekh-Yusef wrote:
>
>> An access token is explicit too; it carries authorization to access some
>> resource/service.
>>
>> I am not sure what you have in mind.
>> Are you suggesting that after the UA/Proxy gets the token from the IdP,
>> that the UA/Proxy would then transform the authorizations given in the
>> token to a SIP caller preference format?
>>
>
> It sounds like you are imagining a situation where the user gets a token
> that authorizes some specific service(s) without the user knowing what
> services those are.


No, that is not what I am proposing.
The user in the conference example will have to explicitly ask for a
specific conference type, e.g. video conference.
There are different ways of obtaining a token with the right authorization
needed for a specific service.



> And then the user makes a request with the token and expects the server to
> figure out what services to supply.
>

Yes, the server providing the service will be able to validate the token,
which explicitly states requested service (e.g. video conference with a
maximum of 5 participants) , and then provide the service accordingly.



>
> That seems backwards to me. I would expect that the user would figure out
> what services he wants, then get some credentials that are sufficient to
> authorize the service, and then request the service and supply the
> credentials to cover it.
>
> It may be that the credentials he has are broad enough to cover lots of
> services. He doesn't want them all, he wants what he wants at the time of
> invocation.
>
> So callerprefs provides a mechanism to express what features you want from
> a callee. To that you then need to supply credentials, in this case in the
> form of an authorization token.
>

Well, if we are sending an authorization token anyway, why would we
separate the token from the authorizations?

Regards,
 Rifaat



>
>         Thanks,
>         Paul
>
> Regards,
>>  Rifaat
>>
>>
>>
>>
>> On Tue, Jun 28, 2016 at 4:55 PM, Paul Kyzivat <pkyzivat@alum.mit.edu
>> <mailto:pkyzivat@alum.mit.edu>> wrote:
>>
>>     On 6/28/16 5:32 AM, Rifaat Shekh-Yusef wrote:
>>
>>             The semantics of the INVITE method are "hey, remote
>>         resource, I'd
>>             like to set up a session with you involving the following
>> media
>>             types." The question of whether or not you accept INVITEs,
>>         which is
>>             to my mind the SIP authorization problem,
>>
>>         This is the main difference between how you and I see this.
>>         I think that the authorization problem is much bigger than just
>>         authorizing a specific SIP method. The authorization should be
>>         dealing
>>         with *access to services* provided by SIP servers regardless of
>>         the SIP
>>         method being used.
>>         Much the same way that HTTP authorizes *access to resources*,
>>         regardless
>>         of the HTTP method being used. You do not need a specific token
>>         for HTTP
>>         GET vs HTTP POST, because the method does not matter; what
>>         matters is
>>         the resource that the method is trying to access.
>>
>>
>>     If you want to tell a server something about what you want it to do,
>>     you can use caller preferences. At least then you are being explicit
>>     about it.
>>
>>     That seems more straightforward than embedding similar information
>>     into an authorization token.
>>
>>             Thanks,
>>             Paul
>>
>>
>>     _______________________________________________
>>     sipcore mailing list
>>     sipcore@ietf.org <mailto:sipcore@ietf.org>
>>     https://www.ietf.org/mailman/listinfo/sipcore
>>
>>
>>
>

--001a113ad63cf4fde2053681d5c2
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Thu, Jun 30, 2016 at 10:43 AM, Paul Kyzivat <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:pkyzivat@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit.=
edu</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-=
left-color:rgb(204,204,204);padding-left:1ex"><span>On 6/30/16 10:29 AM, Ri=
faat Shekh-Yusef wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);p=
adding-left:1ex">
An access token is explicit too; it carries authorization to access some<br=
>
resource/service.<br>
<br>
I am not sure what you have in mind.<br>
Are you suggesting that after the UA/Proxy gets the token from the IdP,<br>
that the UA/Proxy would then transform the authorizations given in the<br>
token to a SIP caller preference format?<br>
</blockquote>
<br></span>
It sounds like you are imagining a situation where the user gets a token th=
at authorizes some specific service(s) without the user knowing what servic=
es those are. </blockquote><div><br></div><div>No, that is not what I am pr=
oposing.=C2=A0</div><div>The user in the conference example will have to ex=
plicitly ask for a specific conference type, e.g. video conference.</div><d=
iv>There are different ways of obtaining a token with the right authorizati=
on needed for a specific service.<br></div><div><br></div><div>=C2=A0</div>=
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);p=
adding-left:1ex">And then the user makes a request with the token and expec=
ts the server to figure out what services to supply.<br></blockquote><div><=
br></div><div>Yes, the server providing the service will be able to validat=
e the token, which explicitly states requested service (e.g. video conferen=
ce with a maximum of 5 participants) , and then provide the service accordi=
ngly.</div><div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote=
" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style=
:solid;border-left-color:rgb(204,204,204);padding-left:1ex">
<br>
That seems backwards to me. I would expect that the user would figure out w=
hat services he wants, then get some credentials that are sufficient to aut=
horize the service, and then request the service and supply the credentials=
 to cover it.<br>
<br>
It may be that the credentials he has are broad enough to cover lots of ser=
vices. He doesn&#39;t want them all, he wants what he wants at the time of =
invocation.<br>
<br>
So callerprefs provides a mechanism to express what features you want from =
a callee. To that you then need to supply credentials, in this case in the =
form of an authorization token.<br></blockquote><div><br></div><div>Well, i=
f we are sending an authorization token anyway, why would we separate the t=
oken from the authorizations?</div><div><br></div><div>Regards,</div><div>=
=C2=A0Rifaat</div><div><br></div><div>=C2=A0</div><blockquote class=3D"gmai=
l_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-lef=
t-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex">
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Thanks,<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Paul<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);p=
adding-left:1ex"><span>
Regards,<br>
=C2=A0Rifaat<br>
<br>
<br>
<br>
<br>
On Tue, Jun 28, 2016 at 4:55 PM, Paul Kyzivat &lt;<a href=3D"mailto:pkyziva=
t@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit.edu</a><br></span><div>=
<div>
&lt;mailto:<a href=3D"mailto:pkyzivat@alum.mit.edu" target=3D"_blank">pkyzi=
vat@alum.mit.edu</a>&gt;&gt; wrote:<br>
<br>
=C2=A0 =C2=A0 On 6/28/16 5:32 AM, Rifaat Shekh-Yusef wrote:<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 The semantics of the INVITE metho=
d are &quot;hey, remote<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 resource, I&#39;d<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 like to set up a session with you=
 involving the following media<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 types.&quot; The question of whet=
her or not you accept INVITEs,<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 which is<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 to my mind the SIP authorization =
problem,<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 This is the main difference between how you and=
 I see this.<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 I think that the authorization problem is much =
bigger than just<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 authorizing a specific SIP method. The authoriz=
ation should be<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 dealing<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 with *access to services* provided by SIP serve=
rs regardless of<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 the SIP<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 method being used.<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Much the same way that HTTP authorizes *access =
to resources*,<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 regardless<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 of the HTTP method being used. You do not need =
a specific token<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 for HTTP<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 GET vs HTTP POST, because the method does not m=
atter; what<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 matters is<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 the resource that the method is trying to acces=
s.<br>
<br>
<br>
=C2=A0 =C2=A0 If you want to tell a server something about what you want it=
 to do,<br>
=C2=A0 =C2=A0 you can use caller preferences. At least then you are being e=
xplicit<br>
=C2=A0 =C2=A0 about it.<br>
<br>
=C2=A0 =C2=A0 That seems more straightforward than embedding similar inform=
ation<br>
=C2=A0 =C2=A0 into an authorization token.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Thanks,<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Paul<br>
<br>
<br>
=C2=A0 =C2=A0 _______________________________________________<br>
=C2=A0 =C2=A0 sipcore mailing list<br></div></div>
=C2=A0 =C2=A0 <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"_bla=
nk">sipcore@ietf.org</a>&gt;<br>
=C2=A0 =C2=A0 <a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" rel=
=3D"noreferrer" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sip=
core</a><br>
<br>
<br>
</blockquote>
<br>
</blockquote></div><br></div></div>

--001a113ad63cf4fde2053681d5c2--


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To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
References: <CAGL6ep+8nhiuQs8uBLxmsdwfrxFys4PM8UP5Apbcb7YVK4fmfg@mail.gmail.com> <D3842CA2.19BA97%jon.peterson@neustar.biz> <CAGL6epKbdjHP-K9Hjn825HUVTQ4V=TnUYeyrLSU0ETQtmsfs9g@mail.gmail.com> <D385862C.19C10C%jon.peterson@neustar.biz> <CAGL6epLZOhUMQHGGW_Y49gZZAdvFWH07NfO97f=krk1_K9Xd-g@mail.gmail.com> <D386CD3B.19C95F%jon.peterson@neustar.biz> <CAGL6epKj00is8dH7NHdjjuTbh-XbeB7yuALouzR2fzfMBC=R4w@mail.gmail.com> <D38F19C3.19F959%jon.peterson@neustar.biz> <CAGL6epK_DEPr5nYOfGOszLtfT8i8Lso8cZkdXO2+v613ix3eDQ@mail.gmail.com> <D3905929.19FF98%jon.peterson@neustar.biz> <CAGL6epKJQqprV7z3GA-PMeTWvzD_CE7=q-rcv9Tyw-WSmHndyA@mail.gmail.com> <D392BAC1.1A0530%jon.peterson@neustar.biz> <CAGL6ep++QG3uKt0Rtib_eXQVqJQS1Amr5njz1os3WT6xDy5T-w@mail.gmail.com> <b8f18bd8-931c-4730-02b0-f36c519d2943@alum.mit.edu> <CAGL6epJYLKNSTa8C8xka47mFot2WES_rEyw_6Shz6RsSkhDX_w@mail.gmail.com> <1a0ef7d8-4fb7-166d-4e9f-25e200eee1e2@alum.mit.edu> <CAGL6epLV+=2+H4MFLs1K2POU09QTmCYmTRsJkn15yKmeDK5M5A@mail.gmail.com>
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Date: Thu, 30 Jun 2016 13:24:17 -0400
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Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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Comment at end

On 6/30/16 1:02 PM, Rifaat Shekh-Yusef wrote:
>
> On Thu, Jun 30, 2016 at 10:43 AM, Paul Kyzivat <pkyzivat@alum.mit.edu
> <mailto:pkyzivat@alum.mit.edu>> wrote:
>
>     On 6/30/16 10:29 AM, Rifaat Shekh-Yusef wrote:
>
>         An access token is explicit too; it carries authorization to
>         access some
>         resource/service.
>
>         I am not sure what you have in mind.
>         Are you suggesting that after the UA/Proxy gets the token from
>         the IdP,
>         that the UA/Proxy would then transform the authorizations given
>         in the
>         token to a SIP caller preference format?
>
>     It sounds like you are imagining a situation where the user gets a
>     token that authorizes some specific service(s) without the user
>     knowing what services those are.
>
> No, that is not what I am proposing.
> The user in the conference example will have to explicitly ask for a
> specific conference type, e.g. video conference.
> There are different ways of obtaining a token with the right
> authorization needed for a specific service.
>
>     And then the user makes a request with the token and expects the
>     server to figure out what services to supply.
>
> Yes, the server providing the service will be able to validate the
> token, which explicitly states requested service (e.g. video conference
> with a maximum of 5 participants) , and then provide the service
> accordingly.
>
>     That seems backwards to me. I would expect that the user would
>     figure out what services he wants, then get some credentials that
>     are sufficient to authorize the service, and then request the
>     service and supply the credentials to cover it.
>
>     It may be that the credentials he has are broad enough to cover lots
>     of services. He doesn't want them all, he wants what he wants at the
>     time of invocation.
>
>     So callerprefs provides a mechanism to express what features you
>     want from a callee. To that you then need to supply credentials, in
>     this case in the form of an authorization token.
>
> Well, if we are sending an authorization token anyway, why would we
> separate the token from the authorizations?

1) Separation of responsibility.

2) What you are suggesting would require authorization servers to have 
the means to dice up authorizations into whatever granularity a user 
wants to request services.

Suppose some systems don't do that. They simply grant you access to a 
conference server that can do all sorts of conferences. But you still 
want to decide whether to have an audio-only conference or an 
audio/video conference. Now you need a different way to request that.

	Thanks,
	Paul


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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: Rifaat Shekh-Yusef <rifaat.ietf@gmail.com>
Thread-Topic: [sipcore] SIP AuthNZ Problem Statement - v3
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Subject: Re: [sipcore] SIP AuthNZ Problem Statement - v3
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But even if we assume a more open environment where the user agent can send=
 INVITEs directly to the conference bridge, all the conference bridge needs=
 is a signature or similar token that indicates the IdP's approval, -not- a=
ny specific attribute.

Of course attributes are needed.
How would otherwise the conference server know which type of conference to =
start (audio, video, etc)? what is the limitation on the number of particip=
ant? etc...

This is exactly what my "deeper level of complexity" in a previous mail pul=
led apart. You need to differentiate a) what kind of conference the user wa=
nts from b) what kind of conference the IdP is willing to authorize.  Your =
token is just an indication of what the IdP is willing to authorize, and if=
 that is a superset of what the user wants, then the IdP will issue a token=
. If it is a subset, then it won't.

So how does the conference server know which type of conference to start? W=
ell, again, you've focused this discussion on the corner case of an ad hoc =
RFC4579 conference, in the ad hoc case the conference bridge has to infer t=
hat from the type of session that the user agent creates with the conferenc=
e bridge with its initial INVITE. It's not really a very good assumption, t=
hough, which is why we quickly run into trouble when we start trying to und=
erstand this corner case. However, I think it would be safe to say that for=
 this ad hoc case, if the IdP was willing to issue an authorization token, =
then it authorizes the media type(s) that the user specified in their INVIT=
E -  and thus that attributes in the token to express what media types the =
IdP allowed would be at best redundant.

But to be clear, a token that conveys what the IdP is willing to authorize =
does not serve the same purpose as a SIP means for a user agent to convey t=
o a conference bridge what kind of conference it would like to have - nor a=
s a means for a user agent to communicate to an IdP what kind of conference=
 it intends to propose. Paul is mentioning mechanisms like caller prefs her=
e because it would be new work, through some SIP mechanism along those line=
s, to convey that information with SIP.

Outside of this corner case, though, the conference URI is acquired in an o=
ut-of-band fashion at the same time as conference policy is set - surely th=
rough HTTP, surely through a web interface, and that's where all the OAuth =
goodness can (and should) play into conference policy management. Trying to=
 inflate this one ad hoc corner case into a murky argument that SIP needs O=
Auth is and has this whole time been a stretch. RFC4353 already says confer=
ence policy management is not a SIP function. I'm not particularly interest=
ed in working on overriding that, since it Is obviously solved by just usin=
g HTTP, without any new work on our part.


The semantics of the INVITE method are "hey, remote resource, I'd like to s=
et up a session with you involving the following media types." The question=
 of whether or not you accept INVITEs, which is to my mind the SIP authoriz=
ation problem,

This is the main difference between how you and I see this.
I think that the authorization problem is much bigger than just authorizing=
 a specific SIP method. The authorization should be dealing with access to =
services provided by SIP servers regardless of the SIP method being used.

Our real disconnect I think is the degree to which you think SIP offers "se=
rvices" or that application control is inside the scope of SIP. It's not. S=
IP is a protocol that allows Internet endpoints to discover one another and=
 to agree on the character of a session they'd like to share, according to =
RFC3261. If I may quote from Section 2  of that document "SIP does not offe=
r conference control services" and indeed "SIP does not provide services." =
 The only thing SIP has to do with conference services is that "SIP can be =
used to initiate a session that uses some other conference control protocol=
", look ahead to BFCP, and of course that SIP user agents can initiate sess=
ions with conference bridges.

Much the same way that HTTP authorizes access to resources, regardless of t=
he HTTP method being used. You do not need a specific token for HTTP GET vs=
 HTTP POST, because the method does not matter; what matters is the resourc=
e that the method is trying to access.

Accessing and changing resources on the Internet is in the scope of HTTP, w=
hich is why HTTP should be used for the purposes you have in mind here. Yes=
, in addition to initiating sessions, SIP has a number of methods that let =
it convey data, such as INFO, MESSAGE, NOTIFY and so on, and in some use ca=
ses, they can similarly provide access to resources. Maybe there's a SIP au=
thorization problem for those methods that we could find. But for the most =
part, the SIP authorization problem should be aligned with the purpose of S=
IP - it should be about deciding whether or not to accept session initiatio=
n attempts. That's a problem I'd be willing to work on.

Jon Peterson
Neustar, Inc.


Regards,
 Rifaat


--_000_D39AC0E11A3271jonpetersonneustarbiz_
Content-Type: text/html; charset="us-ascii"
Content-ID: <899E88441CB5D84F9D870AD3A0BDDF63@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;">
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
<br>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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 even if we assume a more open environment where the user agent can=
 send INVITEs directly to the conference bridge, all the conference bridge =
needs is a signature or similar token that indicates the IdP's approval, -n=
ot- any specific attribute.</div>
</div>
</blockquote>
<div><br>
</div>
<div>Of course attributes are needed.&nbsp;</div>
<div>How would otherwise the conference server know which type of conferenc=
e to start (audio, video, etc)? what is the limitation on the number of par=
ticipant? etc...</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>This is exactly what my &quot;deeper level of complexity&quot; in a pr=
evious mail pulled apart. You need to differentiate a) what kind of confere=
nce the user wants from b) what kind of conference the IdP is willing to au=
thorize. &nbsp;Your token is just an indication
 of what the IdP is willing to authorize, and if that is a superset of what=
 the user wants, then the IdP will issue a token. If it is a subset, then i=
t won't.</div>
<div><br>
</div>
<div>So how does the conference server know which type of conference to sta=
rt? Well, again, you've focused this discussion on the corner case of an ad=
 hoc RFC4579 conference, in the ad hoc case the conference bridge has to in=
fer that from the type of session
 that the user agent creates with the conference bridge with its initial IN=
VITE. It's not really a very good assumption, though, which is why we quick=
ly run into trouble when we start trying to understand this corner case. Ho=
wever, I think it would be safe
 to say that for this ad hoc case, if the IdP was willing to issue an autho=
rization token, then it authorizes the media type(s) that the user specifie=
d in their INVITE - &nbsp;and thus that attributes in the token to express =
what media types the IdP allowed would
 be at best redundant.</div>
<div><br>
</div>
<div>But to be clear, a token that conveys what the IdP is willing to autho=
rize does not serve the same purpose as a SIP means for a user agent to con=
vey to a conference bridge what kind of conference it would like to have - =
nor as a means for a user agent
 to communicate to an IdP what kind of conference it intends to propose. Pa=
ul is mentioning mechanisms like caller prefs here because it would be new =
work, through some SIP mechanism along those lines, to convey that informat=
ion with SIP.</div>
<div><br>
</div>
<div>Outside of this corner case, though, the conference URI is acquired in=
 an out-of-band fashion at the same time as conference policy is set - sure=
ly through HTTP, surely through a web interface, and that's where all the O=
Auth goodness can (and should) play
 into conference policy management. Trying to inflate this one ad hoc corne=
r case into a murky argument that SIP needs OAuth is and has this whole tim=
e been a stretch. RFC4353 already says conference policy management is not =
a SIP function. I'm not particularly
 interested in working on overriding that, since it Is obviously solved by =
just using HTTP, without any new work on our part.</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);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>The semantics of the INVITE method are &quot;hey, remote resource, I'd=
 like to set up a session with you involving the following media types.&quo=
t; The question of whether or not you accept INVITEs, which is to my mind t=
he SIP authorization problem,</div>
</div>
</blockquote>
<div><br>
</div>
<div>This is the main difference between how you and I see this.</div>
<div>I think that the authorization problem is much bigger than just author=
izing a specific SIP method. The authorization should be dealing with
<b>access to services</b> provided by SIP servers regardless of the SIP met=
hod being used.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>Our real disconnect I think is the degree to which you think SIP offer=
s &quot;services&quot; or that application control is inside the scope of S=
IP. It's not. SIP is a protocol that allows Internet endpoints to discover =
one another and to agree on the character
 of a session they'd like to share, according to RFC3261. If I may quote fr=
om Section 2 &nbsp;of that document &quot;SIP does not offer conference con=
trol services&quot; and indeed &quot;SIP does not provide services.&quot; &=
nbsp;The only thing SIP has to do with conference services is
 that &quot;SIP can be used to initiate a session that uses some other conf=
erence control protocol&quot;, look ahead to BFCP, and of course that SIP u=
ser agents can initiate sessions with conference bridges.</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>Much the same way that HTTP authorizes <b>access to resources</b>, reg=
ardless of the HTTP method being used. You do not need a specific token for=
 HTTP GET vs HTTP POST, because the method does not matter; what matters is=
 the resource that the method is
 trying to access.</div>
</div>
</div>
</div>
</blockquote>
</span>
<div><br>
</div>
<div>Accessing and changing resources on the Internet is in the scope of HT=
TP, which is why HTTP should be used for the purposes you have in mind here=
. Yes, in addition to initiating sessions, SIP has a number of methods that=
 let it convey data, such as INFO,
 MESSAGE, NOTIFY and so on, and in some use cases, they can similarly provi=
de access to resources. Maybe there's a SIP authorization problem for those=
 methods that we could find. But for the most part, the SIP authorization p=
roblem should be aligned with the
 purpose of SIP - it should be about deciding whether or not to accept sess=
ion initiation attempts. That's a problem I'd be willing to work on.</div>
<div><br>
</div>
<div>Jon Peterson</div>
<div>Neustar, Inc.</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote id=3D"MAC_OUTLOOK_ATTRIBUTION_BLOCKQUOTE" style=3D"BORDER-LEFT:=
 #b5c4df 5 solid; PADDING:0 0 0 5; MARGIN:0 0 0 5;">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<div>Regards,</div>
<div>&nbsp;Rifaat</div>
<div><br>
</div>
</div>
</div>
</div>
</blockquote>
</span>
</body>
</html>

--_000_D39AC0E11A3271jonpetersonneustarbiz_--


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From: Roman Shpount <roman@telurix.com>
Date: Thu, 30 Jun 2016 18:42:32 -0400
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Subject: Re: [sipcore] Happy Eyeballs: Outbounde
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Content-Type: text/plain; charset=UTF-8

Few observations:

1. There is a significant number of end points, such as Polycom phones,
that do support SIP outbound.

2. The most efficient way for the client to keep the NAT hole open is to
send STUN based keep alive messages. They are widely supported. For
instance OpenSIPS supports via Stun Module.

3. For server, the most efficient way to keep the NAT hole open is to send
OPTIONS or NOTIFY requests to the client. This is much more efficient then
registrations with small timeouts.

4. Registration server which has an in-memory database for current
registrations can be fairly efficient in reducing number of back-end DB
updates due to registrations and subscriptions with small timeouts. It is
not going to be stateless, but its state does not need to be replicated and
would automatically recover on the stand-by server on the next registration
or subscription request from the client.

5. One of the big problems with encrypted SIP traffic is that it gets
modified by ALG. For a lot of hosted PBX providers avoiding the need to
troubleshoot customer routers is a bigger incentive to deploy TLS then user
privacy.

Regards,

_____________
Roman Shpount

On Wed, Jun 29, 2016 at 9:35 PM, Dale R. Worley <worley@ariadne.com> wrote:

> This thread is to explore "outbound" issues.  As I have them, the
> requirements are:
>
>     - Clients behind client-side NATs must be fully supported.
>
>     - The solution should be as compatible with SIP Outbound as
> practicable.
>
>     The difficult part of this is "the solution for simple UDP (i.e., not
>     using SIP Outbound)".  There is a strong perception that in systems
>     where an edge proxy services 10^5 or 10^6 UAs that only
>     UDP-without-Outbound has a low enough per-flow overhead to be
>     workable.  The important requirements for simple UDP seem to be:
>
>     - The per-UA state in the edge proxy must be no larger than about what
>       is kept for a registration.
>
>     - The update rate of the per-UA state in the edge proxy must be no
>       larger than about the update rate for a registration.
>
>     It is perceived that TCP, TLS, and even UDP-with-Outbound do not meet
>     these requirements.  A substantial fraction of the UAs tested in the
>     latest SIPit do not support TCP, TLS, or Outbound, showing that there
>     is substantial market presence of simple-UDP-only UAs.
>
>     It is possible that a "Simple Outbound" can be defined that provides
>     the needed part of the functionality of Outbound at a sufficiently low
>     overhead and complexity that it meets these requirements.  If so, it
>     is desirable that it be conceptually and operationally
>     upward-compatible with Outbound.
>
> It's clear to me that the problem is *solvable*, because existing SIP
> systems do handle the client-side NAT problem.  E.g., the open-source
> sipX system has full client-side NAT support.  That scheme doesn't
> require SIP Outbound support in the client at all.  NAT support is
> triggered by the client's requests arriving from an address that is
> different than what is specified in the request.  Support is
> implemented by manipulating the client's behavior, rewriting
> requests/responses to substitute IP addresses, and providing
> (essentially) a TURN server to relay media.
>
> As far as I can remember, sipX's NAT support is recorded and
> implemented in the standard registration/redirect database.  However,
> NAT support does depend on forcing the client to re-register
> frequently enough to be assured that the NAT mapping is not released.
> Since processing re-registrations is by far the bulk of the signaling
> traffic even without NAT support, this is not a trivial change.
>
> My expectation is that almost all commercial SIP systems have NAT
> support of this sort.
>
> One difference between this sort of NAT support and Outbound is that
> NAT support is done only at the registrar/proxy; if there is a
> separate edge proxy, it only passes UDP messages and can easily be
> stateless.  This might be a significant factor in very large
> deployments.
>
> Perhaps a significant problem with Outbound is that it has to be
> implemented in both the phone and the switch, leading to a network
> effect problem.
>
> At this point, it seems to me that we need to get a better
> understanding of what people are doing in the market to deal with NATs
> and find out why they don't use Outbound.  (Since Outbound is the
> standard method, I would think it has a strategic advantage in the
> technological competition.)
>
> Dale
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

--94eb2c05665cf893ee0536869548
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>Few observations:</div><div><br></div>1. There is a s=
ignificant number of end points, such as Polycom phones, that do support SI=
P outbound.<div><br><div>2. The most efficient way for the client to keep t=
he NAT hole open is to send STUN based keep alive messages. They are widely=
 supported. For instance OpenSIPS supports via Stun Module.</div></div><div=
><br></div><div>3. For server, the most efficient way to keep the NAT hole =
open is to send OPTIONS or NOTIFY requests to the client. This is much more=
 efficient then registrations with small timeouts.</div><div><br></div><div=
>4. Registration server which has an in-memory database for current registr=
ations can be fairly efficient in reducing number of back-end DB updates du=
e to registrations and subscriptions with small timeouts. It is not going t=
o be stateless, but its state does not need to be replicated and would auto=
matically recover on the stand-by server on the next registration or subscr=
iption request from the client.</div><div><br></div><div>5. One of the big =
problems with encrypted SIP traffic is that it gets modified by ALG. For a =
lot of hosted PBX providers avoiding the need to troubleshoot customer rout=
ers is a bigger incentive to deploy TLS then user privacy.</div><div><br></=
div><div>Regards,=C2=A0</div></div><div class=3D"gmail_extra"><br clear=3D"=
all"><div><div class=3D"gmail_signature" data-smartmail=3D"gmail_signature"=
>_____________<br>Roman Shpount</div></div>
<br><div class=3D"gmail_quote">On Wed, Jun 29, 2016 at 9:35 PM, Dale R. Wor=
ley <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"gma=
il_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-lef=
t:1ex">This thread is to explore &quot;outbound&quot; issues.=C2=A0 As I ha=
ve them, the<br>
requirements are:<br>
<br>
=C2=A0 =C2=A0 - Clients behind client-side NATs must be fully supported.<br=
>
<br>
=C2=A0 =C2=A0 - The solution should be as compatible with SIP Outbound as p=
racticable.<br>
<br>
=C2=A0 =C2=A0 The difficult part of this is &quot;the solution for simple U=
DP (i.e., not<br>
=C2=A0 =C2=A0 using SIP Outbound)&quot;.=C2=A0 There is a strong perception=
 that in systems<br>
=C2=A0 =C2=A0 where an edge proxy services 10^5 or 10^6 UAs that only<br>
=C2=A0 =C2=A0 UDP-without-Outbound has a low enough per-flow overhead to be=
<br>
=C2=A0 =C2=A0 workable.=C2=A0 The important requirements for simple UDP see=
m to be:<br>
<br>
=C2=A0 =C2=A0 - The per-UA state in the edge proxy must be no larger than a=
bout what<br>
=C2=A0 =C2=A0 =C2=A0 is kept for a registration.<br>
<br>
=C2=A0 =C2=A0 - The update rate of the per-UA state in the edge proxy must =
be no<br>
=C2=A0 =C2=A0 =C2=A0 larger than about the update rate for a registration.<=
br>
<br>
=C2=A0 =C2=A0 It is perceived that TCP, TLS, and even UDP-with-Outbound do =
not meet<br>
=C2=A0 =C2=A0 these requirements.=C2=A0 A substantial fraction of the UAs t=
ested in the<br>
=C2=A0 =C2=A0 latest SIPit do not support TCP, TLS, or Outbound, showing th=
at there<br>
=C2=A0 =C2=A0 is substantial market presence of simple-UDP-only UAs.<br>
<br>
=C2=A0 =C2=A0 It is possible that a &quot;Simple Outbound&quot; can be defi=
ned that provides<br>
=C2=A0 =C2=A0 the needed part of the functionality of Outbound at a suffici=
ently low<br>
=C2=A0 =C2=A0 overhead and complexity that it meets these requirements.=C2=
=A0 If so, it<br>
=C2=A0 =C2=A0 is desirable that it be conceptually and operationally<br>
=C2=A0 =C2=A0 upward-compatible with Outbound.<br>
<br>
It&#39;s clear to me that the problem is *solvable*, because existing SIP<b=
r>
systems do handle the client-side NAT problem.=C2=A0 E.g., the open-source<=
br>
sipX system has full client-side NAT support.=C2=A0 That scheme doesn&#39;t=
<br>
require SIP Outbound support in the client at all.=C2=A0 NAT support is<br>
triggered by the client&#39;s requests arriving from an address that is<br>
different than what is specified in the request.=C2=A0 Support is<br>
implemented by manipulating the client&#39;s behavior, rewriting<br>
requests/responses to substitute IP addresses, and providing<br>
(essentially) a TURN server to relay media.<br>
<br>
As far as I can remember, sipX&#39;s NAT support is recorded and<br>
implemented in the standard registration/redirect database.=C2=A0 However,<=
br>
NAT support does depend on forcing the client to re-register<br>
frequently enough to be assured that the NAT mapping is not released.<br>
Since processing re-registrations is by far the bulk of the signaling<br>
traffic even without NAT support, this is not a trivial change.<br>
<br>
My expectation is that almost all commercial SIP systems have NAT<br>
support of this sort.<br>
<br>
One difference between this sort of NAT support and Outbound is that<br>
NAT support is done only at the registrar/proxy; if there is a<br>
separate edge proxy, it only passes UDP messages and can easily be<br>
stateless.=C2=A0 This might be a significant factor in very large<br>
deployments.<br>
<br>
Perhaps a significant problem with Outbound is that it has to be<br>
implemented in both the phone and the switch, leading to a network<br>
effect problem.<br>
<br>
At this point, it seems to me that we need to get a better<br>
understanding of what people are doing in the market to deal with NATs<br>
and find out why they don&#39;t use Outbound.=C2=A0 (Since Outbound is the<=
br>
standard method, I would think it has a strategic advantage in the<br>
technological competition.)<br>
<br>
Dale<br>
<br>
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</blockquote></div><br></div>

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