
From spromano@unina.it  Fri Jun  3 11:41:42 2011
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Date: Fri, 03 Jun 2011 20:41:29 +0200
From: Simon Pietro Romano <spromano@unina.it>
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Subject: [splices] Agenda request for IETF 81
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Hi all,

I just requested a session (1 hour long) for a f2f meeting in Quebec 
City. Please, send me your requests to reserve a presentation slot 
during the meeting asap.

Cheers,

Simon

-- 
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                     Simon Pietro Romano
               Universita' di Napoli Federico II
                  Computer Science Department
         Phone: +39 081 7683823 -- Fax: +39 081 7684219
                 e-mail: spromano@unina.it
           http://www.comics.unina.it/simonpietro.romano

     <<Molti mi dicono che lo scoraggiamento è l'alibi degli
    idioti. Ci rifletto un istante; e mi scoraggio>>. Magritte.
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From musgravepj@gmail.com  Sun Jun  5 12:15:15 2011
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Date: Sun, 5 Jun 2011 15:15:10 -0400
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From: Peter Musgrave <musgravepj@gmail.com>
To: "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>
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Subject: Re: [splices] Using Two Separate Devices to Start a Conversation proposal
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HI Rifaat,

I have concerns about this approach. Fundamentally the RTP stream is not
symmetric (in the sense of RFC4961). This has consequences for NAT
traversal, general interoperability and media path security schemes like
ZRTP.

I think an approach in which one of the loosely coupled endpoints acts as a
media relay would eliminate all these issues. It can then source and sink
audio on the same port to the external device and provide the appearance of
symmetric RTP. (This approach would also have merit in the case where one LC
endpoint does audio and the other does video since there is now an RTP/RTCP
path between the endpoints for exchanging data for the purposes of
lipsync).

I recognize this imposes an additional role of relaying media in the
implementation of a LC device - but if the objective is to be interoperable
with non LC devices "in the wild" - I see no other alternative.

Regards,

Peter

On Thu, May 26, 2011 at 8:37 PM, Shekh-Yusef, Rifaat (Rifaat) <
rifatyu@avaya.com> wrote:

> Hi,
>
> The following proposal is to cover the first use case in the disaggregated
> media draft.
>
> Regards,
>  Rifaat
>
>
>   Alice                 Alice                Proxy                  Bob
>    PC                 Desk Phone
>    |                     |                     |                     |
>    |                     |                     |                     |
> Both Alice's devices subscribe to the reg event package, which allows each
> device to
> discover the capabilities of the other device based on the feature tags
> provided by each device.
> The Desk Phone knows that the PC supports Video, while the PC knows that
> the Desk Phone only supports audio.
>    |                     |                     |                     |
>    |                     | SUBSCRIBE reg       |                     |
>    |                     |-------------------->|                     |
>    |                     | 200 OK              |                     |
>    |                     |<--------------------|                     |
>    |                     |                     |                     |
>    | SUBSCRIBE reg       |                     |                     |
>    |------------------------------------------>|                     |
>    | 200 OK              |                     |                     |
>    |<------------------------------------------|                     |
>    |                     |                     |                     |
>    |                     |                     |                     |
> The two devices also subscribe to the dialog of each other.
>    |                     |                     |                     |
>    |                     |                     |                     |
>    | SUBSCRIBE dialog    |                     |                     |
>    |-------------------->|                     |                     |
>    | 200 OK              |                     |                     |
>    |<--------------------|                     |                     |
>    |                     |                     |                     |
>    | SUBSCRIBE dialog    |                     |                     |
>    |<--------------------|                     |                     |
>    | 200 OK              |                     |                     |
>    |-------------------->|                     |                     |
>    |                     |                     |                     |
>    |                     |                     |                     |
>
>
>
> Alice wants to initiate a call to Bob, but wants to use her Desk Phone to
> receive the audio from Bob and use her PC to send audio to Bob.
> The PC instructs the Desk Phone to initiate a call to Bob in a receive only
> mode.
>
>    |                     |                     |                     |
>    | INVOKE Action:
> urn:invoke:call:initiate;media=audio;direction=recvonly;target=bob@whatever
>    |-------------------->|                     |                     |
>    | OK                  |                     |                     |
>    |<--------------------|                     |                     |
>    |                     | INVITE [recvonly]   |                     |
>    |                     |-------------------->|                     |
>    |                     |                     | INVITE [recvonly]   |
>    |                     |                     |-------------------->|
>    |                     |                     | 180                 |
>    |                     | 180                 |<--------------------|
>    |                     |<--------------------|                     |
>    |                     |                     | 200 OK              |
>    |                     | 200 OK              |<--------------------|
>    |                     |<--------------------|                     |
>    |                     |                     |                     |
> As soon as Bob answers the call, a NOTIFY is sent to Alice's PC (dialog
> event package)
>    |                     |                     |                     |
>    | NOTIFY              |                     |                     |
>    |<--------------------|                     |                     |
>    | OK                  |                     |                     |
>    |-------------------->|                     |                     |
>    |                     |                     |                     |
> PC sends a new INVITE with Join with a sendonly SDP offer
>    |                     |                     |                     |
>    | INVITE Join [sendonly]                    |                     |
>    |-------------------->|                     |                     |
>    | 100 Trying          |                     |                     |
>    |<--------------------|                     |                     |
>    |                     |                     |                     |
>    |                     | re-INVITE [sendrecv]|                     |
>    |                     |-------------------->|                     |
>    |                     |                     | re-INVITE [sendrecv]|
>    |                     |                     |-------------------->|
>    |                     |                     | 200 OK              |
>    |                     | 200 OK              |<--------------------|
>    |                     |<--------------------|                     |
>    | 200 OK              |                     |                     |
>    |<--------------------|                     |                     |
>    |                     |                     |                     |
>    |                     |                     |                     |
>    |<----dialog2-------->|<---dialog1------------------------------->|
>    |                     |                     |                     |
>    |                     |<======audio===============================|
>    |=============================audio==============================>|
>    |                     |                     |                     |
>    |                     |                     |                     |
>
> _______________________________________________
> splices mailing list
> splices@ietf.org
> https://www.ietf.org/mailman/listinfo/splices
>

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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: Peter Musgrave <musgravepj@gmail.com>, "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>
Date: Mon, 6 Jun 2011 15:07:29 -0400
Thread-Topic: [splices] Using Two Separate Devices to Start a Conversation proposal
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Subject: Re: [splices] Using Two Separate Devices to Start a Conversation	proposal
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________________________________________
From: splices-bounces@ietf.org [splices-bounces@ietf.org] On Behalf Of Pete=
r Musgrave [musgravepj@gmail.com]

I have concerns about this approach. Fundamentally the RTP stream is not sy=
mmetric (in the sense of RFC4961). This has consequences for NAT traversal,=
 general interoperability and media path security schemes like ZRTP.
_______________________________________________

I have to agree and disagree...  As Rifaat diagrammed it, the call should w=
ork, as it uses already-defined SIP facilities.  Indeed, it's rather clever=
, I'd never thought of joining a sendonly dialog with a recvonly dialog.  T=
here are some limitations, in that we depend on the far-end UA to execute t=
he join in the way we want.

The problems you raise, those of asymmetric RTP, already exist -- One can m=
ake a sendonly audio call, or a recvonly audio call.  We need to ensure tha=
t NAT traversal and ZRTP work correctly when  media stream is set up to be =
one-way.

Dale


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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: Peter Musgrave <musgravepj@gmail.com>, "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>
Date: Mon, 6 Jun 2011 15:19:49 -0400
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________________________________________
From: splices-bounces@ietf.org [splices-bounces@ietf.org] On Behalf Of Pete=
r Musgrave [musgravepj@gmail.com]

I think an approach in which one of the loosely coupled endpoints acts as a=
 media relay would eliminate all these issues. It can then source and sink =
audio on the same port to the external device and provide the appearance of=
 symmetric RTP.
_______________________________________________

I like the approach of having an aggregation controller, probably incorpora=
ted into one of the UAs, that coordinates the other aggregated UAs using 3P=
CC techniques.  (Either relaying the RTP, or manipulating the SDP to connec=
t each source and sink.)

Of course, the aggregation controller has to handle all the "macro" call-co=
ntrol actions, such as receiving and acting on REFER, sending REFER, receiv=
ing and acting on INVITE-Replaces and INVITE-Join, etc.  Interestingly, all=
 of the proposals require that one UA or the other handle all of these circ=
umstances, although with a 3PCC aggregation controller, the need for design=
ing how to handle these actions is more obvious.

The one additional operation is how does an aggregation controller hand ove=
r a call to another aggregation controller?  The aggregation process contai=
ns quite a bit of state (although some of it is reported in the dialog even=
ts of the aggregation controller).

BTW, here is a case which shows why it is better to have  the aggregation p=
rocess done entirely at one end of the call:  Consider two aggregated clust=
ers of UAs maintaining a dialog between themselves.  If a UA had to be awar=
e of the aggregation at the far end of the call, along with the aggregation=
 it was participating in on the near end, the logic would be incredibly com=
plicated.

Dale

From alan.b.johnston@gmail.com  Mon Jun  6 12:36:07 2011
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From: Alan Johnston <alan.b.johnston@gmail.com>
Date: Mon, 6 Jun 2011 14:35:56 -0500
To: "Worley, Dale R (Dale)" <dworley@avaya.com>
Cc: "splices@ietf.org" <splices@ietf.org>
Subject: Re: [splices] Using Two Separate Devices to Start a Conversation proposal
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If I understand correctly, there will be two separate RTP streams, two m=3D l=
ines?

If this is the case, then two separate NAT traversal mechanisms will be used=
 (i.e. ICE rubs twice or two relays used). As far as ZRTP or other media pat=
h keying protocols, each session will be keyed separately. With ZRTP the end=
point will see two different ZIDs.  This does unfortunately mean two Diffie H=
ellman calculations.=20

Since these are separate sessions, each will have an RTCP session as well, a=
nd these may need NAT traversal as well.

So this will work, but things like logging and quality reporting will be qui=
te complicated.=20

- Alan -



On Jun 6, 2011, at 2:07 PM, "Worley, Dale R (Dale)" <dworley@avaya.com> wrot=
e:

> ________________________________________
> From: splices-bounces@ietf.org [splices-bounces@ietf.org] On Behalf Of Pet=
er Musgrave [musgravepj@gmail.com]
>=20
> I have concerns about this approach. Fundamentally the RTP stream is not s=
ymmetric (in the sense of RFC4961). This has consequences for NAT traversal,=
 general interoperability and media path security schemes like ZRTP.
> _______________________________________________
>=20
> I have to agree and disagree...  As Rifaat diagrammed it, the call should w=
ork, as it uses already-defined SIP facilities.  Indeed, it's rather clever,=
 I'd never thought of joining a sendonly dialog with a recvonly dialog.  The=
re are some limitations, in that we depend on the far-end UA to execute the j=
oin in the way we want.
>=20
> The problems you raise, those of asymmetric RTP, already exist -- One can m=
ake a sendonly audio call, or a recvonly audio call.  We need to ensure that=
 NAT traversal and ZRTP work correctly when  media stream is set up to be on=
e-way.
>=20
> Dale
>=20
> _______________________________________________
> splices mailing list
> splices@ietf.org
> https://www.ietf.org/mailman/listinfo/splices

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Subject: Re: [splices] Using Two Separate Devices to Start a	Conversation proposal
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IMO it is *undesirable* to require that there be a common point through 
which all the media to/from that end must flow. Its especially 
problematic if the "intelligent" device that would have the capability 
to do this is bandwidth limited - as could be the case with mobile phones.

Maybe this can be tolerated if we can find nothing better. Or maybe it 
can be a fallback approach if something else doesn't work. But I'd hate 
to see defining that as *the* approach without trying to find a way to 
avoid it.

	Thanks,
	Paul

On 6/6/2011 3:19 PM, Worley, Dale R (Dale) wrote:
> ________________________________________
> From: splices-bounces@ietf.org [splices-bounces@ietf.org] On Behalf Of Peter Musgrave [musgravepj@gmail.com]
>
> I think an approach in which one of the loosely coupled endpoints acts as a media relay would eliminate all these issues. It can then source and sink audio on the same port to the external device and provide the appearance of symmetric RTP.
> _______________________________________________
>
> I like the approach of having an aggregation controller, probably incorporated into one of the UAs, that coordinates the other aggregated UAs using 3PCC techniques.  (Either relaying the RTP, or manipulating the SDP to connect each source and sink.)
>
> Of course, the aggregation controller has to handle all the "macro" call-control actions, such as receiving and acting on REFER, sending REFER, receiving and acting on INVITE-Replaces and INVITE-Join, etc.  Interestingly, all of the proposals require that one UA or the other handle all of these circumstances, although with a 3PCC aggregation controller, the need for designing how to handle these actions is more obvious.
>
> The one additional operation is how does an aggregation controller hand over a call to another aggregation controller?  The aggregation process contains quite a bit of state (although some of it is reported in the dialog events of the aggregation controller).
>
> BTW, here is a case which shows why it is better to have  the aggregation process done entirely at one end of the call:  Consider two aggregated clusters of UAs maintaining a dialog between themselves.  If a UA had to be aware of the aggregation at the far end of the call, along with the aggregation it was participating in on the near end, the logic would be incredibly complicated.
>
> Dale
> _______________________________________________
> splices mailing list
> splices@ietf.org
> https://www.ietf.org/mailman/listinfo/splices
>

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Subject: Re: [splices] Using Two Separate Devices to Start a Conversation proposal
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On 6/6/2011 3:35 PM, Alan Johnston wrote:
> If I understand correctly, there will be two separate RTP streams, two m= lines?
>
> If this is the case, then two separate NAT traversal mechanisms will be used (i.e. ICE rubs twice or two relays used). As far as ZRTP or other media path keying protocols, each session will be keyed separately. With ZRTP the endpoint will see two different ZIDs.  This does unfortunately mean two Diffie Hellman calculations.
>
> Since these are separate sessions, each will have an RTCP session as well, and these may need NAT traversal as well.
>
> So this will work, but things like logging and quality reporting will be quite complicated.

As Dale mentioned, this also depends on the other end doing the "right 
thing" with this case. It is a leap of faith to assume it will realize 
it should accept both streams and use one for input and the other for 
output.

ISTM its at least as likely that it would accept the first stream, 
reject the second, and then have only a one-way "conversation".

	Thanks,
	Paul

> - Alan -
>
>
>
> On Jun 6, 2011, at 2:07 PM, "Worley, Dale R (Dale)"<dworley@avaya.com>  wrote:
>
>> ________________________________________
>> From: splices-bounces@ietf.org [splices-bounces@ietf.org] On Behalf Of Peter Musgrave [musgravepj@gmail.com]
>>
>> I have concerns about this approach. Fundamentally the RTP stream is not symmetric (in the sense of RFC4961). This has consequences for NAT traversal, general interoperability and media path security schemes like ZRTP.
>> _______________________________________________
>>
>> I have to agree and disagree...  As Rifaat diagrammed it, the call should work, as it uses already-defined SIP facilities.  Indeed, it's rather clever, I'd never thought of joining a sendonly dialog with a recvonly dialog.  There are some limitations, in that we depend on the far-end UA to execute the join in the way we want.
>>
>> The problems you raise, those of asymmetric RTP, already exist -- One can make a sendonly audio call, or a recvonly audio call.  We need to ensure that NAT traversal and ZRTP work correctly when  media stream is set up to be one-way.
>>
>> Dale
>>
>> _______________________________________________
>> splices mailing list
>> splices@ietf.org
>> https://www.ietf.org/mailman/listinfo/splices
> _______________________________________________
> splices mailing list
> splices@ietf.org
> https://www.ietf.org/mailman/listinfo/splices
>

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To: Paul Kyzivat <pkyzivat@cisco.com>, "splices@ietf.org" <splices@ietf.org>
Date: Mon, 6 Jun 2011 15:45:49 -0400
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Subject: Re: [splices] Using Two Separate Devices to Start a	Conversation proposal
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________________________________________
From: splices-bounces@ietf.org [splices-bounces@ietf.org] On Behalf Of Paul=
 Kyzivat [pkyzivat@cisco.com]

IMO it is *undesirable* to require that there be a common point through
which all the media to/from that end must flow. Its especially
problematic if the "intelligent" device that would have the capability
to do this is bandwidth limited - as could be the case with mobile phones.
________________________________________

A good point -- We've been assuming that all of the aggregated UAs are near
each other in network terms.  But this use-case shows that to be false:  th=
e controller
is my smartphone, but the video source/sink is my PC.

Of course we can accommodate this within a 3PCC environment by having the c=
ontroller
shuffle SDP information to connect sources/sinks, rather than relaying RTP.

Dale

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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: Alan Johnston <alan.b.johnston@gmail.com>
Date: Mon, 6 Jun 2011 15:49:12 -0400
Thread-Topic: [splices] Using Two Separate Devices to Start a Conversation proposal
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Subject: Re: [splices] Using Two Separate Devices to Start a Conversation proposal
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> If I understand correctly, there will be two separate RTP streams,
> two m=3D lines?

There will be two m=3D lines, but it is one m=3D line in one dialog, and
another m=3D line in the other dialog.

> If this is the case, then two separate NAT traversal mechanisms will
> be used (i.e. ICE rubs twice or two relays used). As far as ZRTP or
> other media path keying protocols, each session will be keyed
> separately. With ZRTP the endpoint will see two different ZIDs.
> This does unfortunately mean two Diffie Hellman calculations.

I think the envisioned problem is with simpler NAT traversal
mechanisms which do not use ICE, but rather depend on the RTP packets
traveling "out" through the NAT to establish and maintain an inward
mapping "pinhole".  That won't work with one-way media, or rather,
inward-only media.

> Since these are separate sessions, each will have an RTCP session as
> well, and these may need NAT traversal as well.

> So this will work, but things like logging and quality reporting
> will be quite complicated.

Hmmm...  In this case, they won't be any harder than they would be in
a three-way ad-hoc conference call.

Dale

From alan.b.johnston@gmail.com  Mon Jun  6 13:34:31 2011
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Subject: Re: [splices] Using Two Separate Devices to Start a Conversation proposal
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On Mon, Jun 6, 2011 at 2:48 PM, Paul Kyzivat <pkyzivat@cisco.com> wrote:
>
>
> On 6/6/2011 3:35 PM, Alan Johnston wrote:
>>
>> If I understand correctly, there will be two separate RTP streams, two m=
=3D
>> lines?
>>
>> If this is the case, then two separate NAT traversal mechanisms will be
>> used (i.e. ICE rubs twice or two relays used). As far as ZRTP or other m=
edia
>> path keying protocols, each session will be keyed separately. With ZRTP =
the
>> endpoint will see two different ZIDs. =A0This does unfortunately mean tw=
o
>> Diffie Hellman calculations.
>>
>> Since these are separate sessions, each will have an RTCP session as wel=
l,
>> and these may need NAT traversal as well.
>>
>> So this will work, but things like logging and quality reporting will be
>> quite complicated.
>
> As Dale mentioned, this also depends on the other end doing the "right
> thing" with this case. It is a leap of faith to assume it will realize it
> should accept both streams and use one for input and the other for output=
.
>
> ISTM its at least as likely that it would accept the first stream, reject
> the second, and then have only a one-way "conversation".

OK, I understand the problem right now.  The problem is that this is a
special kind of "Join" operation.  It is kind of a join, but not
quite.

The operation we are after is perhaps similar to 3.3.8 Far-Fork in RFC 5850=
:

I suspect instead of Join, we need a new primitive, lets call it
Splice or Merge which requests the UAS to add the media in the INVITE
into the dialog that is referenced.

- Alan -

>
> =A0 =A0 =A0 =A0Thanks,
> =A0 =A0 =A0 =A0Paul
>
>> - Alan -
>>
>>
>>
>> On Jun 6, 2011, at 2:07 PM, "Worley, Dale R (Dale)"<dworley@avaya.com>
>> =A0wrote:
>>
>>> ________________________________________
>>> From: splices-bounces@ietf.org [splices-bounces@ietf.org] On Behalf Of
>>> Peter Musgrave [musgravepj@gmail.com]
>>>
>>> I have concerns about this approach. Fundamentally the RTP stream is no=
t
>>> symmetric (in the sense of RFC4961). This has consequences for NAT
>>> traversal, general interoperability and media path security schemes lik=
e
>>> ZRTP.
>>> _______________________________________________
>>>
>>> I have to agree and disagree... =A0As Rifaat diagrammed it, the call sh=
ould
>>> work, as it uses already-defined SIP facilities. =A0Indeed, it's rather
>>> clever, I'd never thought of joining a sendonly dialog with a recvonly
>>> dialog. =A0There are some limitations, in that we depend on the far-end=
 UA to
>>> execute the join in the way we want.
>>>
>>> The problems you raise, those of asymmetric RTP, already exist -- One c=
an
>>> make a sendonly audio call, or a recvonly audio call. =A0We need to ens=
ure
>>> that NAT traversal and ZRTP work correctly when =A0media stream is set =
up to
>>> be one-way.
>>>
>>> Dale
>>>
>>> _______________________________________________
>>> splices mailing list
>>> splices@ietf.org
>>> https://www.ietf.org/mailman/listinfo/splices
>>
>> _______________________________________________
>> splices mailing list
>> splices@ietf.org
>> https://www.ietf.org/mailman/listinfo/splices
>>
> _______________________________________________
> splices mailing list
> splices@ietf.org
> https://www.ietf.org/mailman/listinfo/splices
>

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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: Alan Johnston <alan.b.johnston@gmail.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Mon, 6 Jun 2011 17:15:57 -0400
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Subject: Re: [splices] Using Two Separate Devices to Start a Conversation	proposal
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No, surely it is an ordinary join.  At the far UA, there are two incoming d=
ialogs, each with one audio m=3D line.  The only oddity is that one dialog =
is recvonly (at the far UA), and so is never sent media.  The other dialog =
is sendonly (at the far UA) and so never provides audio to the mixing, but =
is only sent the result of the mixing.  The effect is that the far UA's inp=
ut audio is sent only to the second dialog, and the far UA's output audio i=
s driven only by the first dialog.

Dale

From alan.b.johnston@gmail.com  Mon Jun  6 14:45:42 2011
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But isn't the INVITE Join received at Alice's Desk Phone?  If so, then
a normal Join will result in Alice's Desk Phone performing mixing of
audio streams from Alice's PC, Alice's Desk Phone, and Bob.  I thought
what was desired (and shown by Rifaat's flow) was that Alice's Desk
phone did not mix the media, but instead inserted/spliced the media
from Alice's PC into the dialog with Bob, resulting in a re-INVITE to
Bob with two m=3D lines.

- Alan -

On Mon, Jun 6, 2011 at 4:15 PM, Worley, Dale R (Dale) <dworley@avaya.com> w=
rote:
> No, surely it is an ordinary join. =A0At the far UA, there are two incomi=
ng dialogs, each with one audio m=3D line. =A0The only oddity is that one d=
ialog is recvonly (at the far UA), and so is never sent media. =A0The other=
 dialog is sendonly (at the far UA) and so never provides audio to the mixi=
ng, but is only sent the result of the mixing. =A0The effect is that the fa=
r UA's input audio is sent only to the second dialog, and the far UA's outp=
ut audio is driven only by the first dialog.
>
> Dale
>

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Subject: Re: [splices] Using Two Separate Devices to Start a Conversation proposal
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OK, forget what I said about this. I didn't read the details and made a 
bad assumption about them.

	Sorry,
	Paul

On 6/6/2011 5:15 PM, Worley, Dale R (Dale) wrote:
> No, surely it is an ordinary join.  At the far UA, there are two incoming dialogs, each with one audio m= line.  The only oddity is that one dialog is recvonly (at the far UA), and so is never sent media.  The other dialog is sendonly (at the far UA) and so never provides audio to the mixing, but is only sent the result of the mixing.  The effect is that the far UA's input audio is sent only to the second dialog, and the far UA's output audio is driven only by the first dialog.
>
> Dale
>

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--0023545309701df23c04a513744a
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Hi all,

My assumptions were that:
- Alice's desk phone is doing the media aggregation and creating a reINVITE
for Bob with a=sendrecv and asymmetric media:
  * media is sent from Bob to Alice desk phone
  * media is sent from Alice Computer to Bob

As far as Bob knows this is a dialog only with Alice's deskphone. Bob does
not have to support SPLICES or do anything special.

Bob must handle asymmetric RTP from Alice.

Ok?

I agree that requiring media relay does become a bad idea when the relaying
element is e.g a cellphone.

Do we think that having asymmetric RTP will be ok? A lot of implementation
have worked toward symmetric RTP (and some will only play RTP arriving from
the place they are sending to! [1 implementation at SIPIT28, 3 at SIPIT27 -
so maybe not worth worrying about]). Are there any issues with ZRTP or use
of ICE? [I am not an ICE expert so I can't recall if symmetric RTP was baked
into ICE, umm "frozen in" to ICE maybe?]

Regards,

Peter





On Mon, Jun 6, 2011 at 5:45 PM, Alan Johnston <alan.b.johnston@gmail.com>wrote:

> But isn't the INVITE Join received at Alice's Desk Phone?  If so, then
> a normal Join will result in Alice's Desk Phone performing mixing of
> audio streams from Alice's PC, Alice's Desk Phone, and Bob.  I thought
> what was desired (and shown by Rifaat's flow) was that Alice's Desk
> phone did not mix the media, but instead inserted/spliced the media
> from Alice's PC into the dialog with Bob, resulting in a re-INVITE to
> Bob with two m= lines.
>
> - Alan -
>
> On Mon, Jun 6, 2011 at 4:15 PM, Worley, Dale R (Dale) <dworley@avaya.com>
> wrote:
> > No, surely it is an ordinary join.  At the far UA, there are two incoming
> dialogs, each with one audio m= line.  The only oddity is that one dialog is
> recvonly (at the far UA), and so is never sent media.  The other dialog is
> sendonly (at the far UA) and so never provides audio to the mixing, but is
> only sent the result of the mixing.  The effect is that the far UA's input
> audio is sent only to the second dialog, and the far UA's output audio is
> driven only by the first dialog.
> >
> > Dale
> >
> _______________________________________________
> splices mailing list
> splices@ietf.org
> https://www.ietf.org/mailman/listinfo/splices
>

--0023545309701df23c04a513744a
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Hi all,=A0<div><br></div><div>My assumptions were that:</div><div>- Alice&#=
39;s desk phone is doing the media aggregation and creating a reINVITE for =
Bob with a=3Dsendrecv and asymmetric media:</div><div>=A0 * media is sent f=
rom Bob to Alice desk phone</div>
<div>=A0 * media is sent from Alice Computer to Bob</div><div><br></div><di=
v>As far as Bob knows this is a dialog only with Alice&#39;s deskphone. Bob=
 does not have to support SPLICES or do anything special.=A0</div><div><br>
</div><div>Bob must handle asymmetric RTP from Alice.</div><div><br></div><=
div>Ok?</div><div><br></div><div>I agree that requiring media relay does be=
come a bad idea when the relaying element is e.g a cellphone.=A0</div><div>
<br></div><div>Do we think that having asymmetric RTP will be ok? A lot of =
implementation have worked toward symmetric RTP (and some will only play RT=
P arriving from the place they are sending to! [1 implementation at SIPIT28=
, 3 at SIPIT27 - so maybe not worth worrying about]). Are there any issues =
with ZRTP or use of ICE? [I am not an ICE expert so I can&#39;t recall if s=
ymmetric RTP was baked into ICE, umm &quot;frozen in&quot; to ICE maybe?]</=
div>
<div><br></div><div>Regards,=A0</div><div><br></div><div>Peter</div><div><b=
r></div><div><br></div><div><br></div><div><br></div><div><br><div class=3D=
"gmail_quote">On Mon, Jun 6, 2011 at 5:45 PM, Alan Johnston <span dir=3D"lt=
r">&lt;<a href=3D"mailto:alan.b.johnston@gmail.com">alan.b.johnston@gmail.c=
om</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;">But isn&#39;t the INVITE Join received at A=
lice&#39;s Desk Phone? =A0If so, then<br>
a normal Join will result in Alice&#39;s Desk Phone performing mixing of<br=
>
audio streams from Alice&#39;s PC, Alice&#39;s Desk Phone, and Bob. =A0I th=
ought<br>
what was desired (and shown by Rifaat&#39;s flow) was that Alice&#39;s Desk=
<br>
phone did not mix the media, but instead inserted/spliced the media<br>
from Alice&#39;s PC into the dialog with Bob, resulting in a re-INVITE to<b=
r>
Bob with two m=3D lines.<br>
<font color=3D"#888888"><br>
- Alan -<br>
</font><div><div></div><div class=3D"h5"><br>
On Mon, Jun 6, 2011 at 4:15 PM, Worley, Dale R (Dale) &lt;<a href=3D"mailto=
:dworley@avaya.com">dworley@avaya.com</a>&gt; wrote:<br>
&gt; No, surely it is an ordinary join. =A0At the far UA, there are two inc=
oming dialogs, each with one audio m=3D line. =A0The only oddity is that on=
e dialog is recvonly (at the far UA), and so is never sent media. =A0The ot=
her dialog is sendonly (at the far UA) and so never provides audio to the m=
ixing, but is only sent the result of the mixing. =A0The effect is that the=
 far UA&#39;s input audio is sent only to the second dialog, and the far UA=
&#39;s output audio is driven only by the first dialog.<br>

&gt;<br>
&gt; Dale<br>
&gt;<br>
_______________________________________________<br>
splices mailing list<br>
<a href=3D"mailto:splices@ietf.org">splices@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/splices" target=3D"_blank"=
>https://www.ietf.org/mailman/listinfo/splices</a><br>
</div></div></blockquote></div><br></div>

--0023545309701df23c04a513744a--

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Subject: Re: [splices] Using Two Separate Devices to Start a Conversation proposal
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I suspect asymmetric rtp *won't* work a lot of the time, for many 
reasons, whether we think they are reasonable or not.

Maybe this is just an unreasonable use case. I can't really imagine 
wanting to do this. If I have a handset, or headset, with a mic, it 
probably has a "speaker" too, and it would seem weird using the mic and 
not its speaker too. And when they are separate there is likely to be 
problems with echos.

	Thanks,
	Paul

On 6/6/2011 7:27 PM, Peter Musgrave wrote:
> Hi all,
>
> My assumptions were that:
> - Alice's desk phone is doing the media aggregation and creating a
> reINVITE for Bob with a=sendrecv and asymmetric media:
>    * media is sent from Bob to Alice desk phone
>    * media is sent from Alice Computer to Bob
>
> As far as Bob knows this is a dialog only with Alice's deskphone. Bob
> does not have to support SPLICES or do anything special.
>
> Bob must handle asymmetric RTP from Alice.
>
> Ok?
>
> I agree that requiring media relay does become a bad idea when the
> relaying element is e.g a cellphone.
>
> Do we think that having asymmetric RTP will be ok? A lot of
> implementation have worked toward symmetric RTP (and some will only play
> RTP arriving from the place they are sending to! [1 implementation at
> SIPIT28, 3 at SIPIT27 - so maybe not worth worrying about]). Are there
> any issues with ZRTP or use of ICE? [I am not an ICE expert so I can't
> recall if symmetric RTP was baked into ICE, umm "frozen in" to ICE maybe?]
>
> Regards,
>
> Peter
>
>
>
>
>
> On Mon, Jun 6, 2011 at 5:45 PM, Alan Johnston <alan.b.johnston@gmail.com
> <mailto:alan.b.johnston@gmail.com>> wrote:
>
>     But isn't the INVITE Join received at Alice's Desk Phone?  If so, then
>     a normal Join will result in Alice's Desk Phone performing mixing of
>     audio streams from Alice's PC, Alice's Desk Phone, and Bob.  I thought
>     what was desired (and shown by Rifaat's flow) was that Alice's Desk
>     phone did not mix the media, but instead inserted/spliced the media
>     from Alice's PC into the dialog with Bob, resulting in a re-INVITE to
>     Bob with two m= lines.
>
>     - Alan -
>
>     On Mon, Jun 6, 2011 at 4:15 PM, Worley, Dale R (Dale)
>     <dworley@avaya.com <mailto:dworley@avaya.com>> wrote:
>      > No, surely it is an ordinary join.  At the far UA, there are two
>     incoming dialogs, each with one audio m= line.  The only oddity is
>     that one dialog is recvonly (at the far UA), and so is never sent
>     media.  The other dialog is sendonly (at the far UA) and so never
>     provides audio to the mixing, but is only sent the result of the
>     mixing.  The effect is that the far UA's input audio is sent only to
>     the second dialog, and the far UA's output audio is driven only by
>     the first dialog.
>      >
>      > Dale
>      >
>     _______________________________________________
>     splices mailing list
>     splices@ietf.org <mailto:splices@ietf.org>
>     https://www.ietf.org/mailman/listinfo/splices
>
>
>
>
> _______________________________________________
> splices mailing list
> splices@ietf.org
> https://www.ietf.org/mailman/listinfo/splices

From rifatyu@avaya.com  Mon Jun  6 19:02:57 2011
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From: "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, "splices@ietf.org" <splices@ietf.org>
Date: Mon, 6 Jun 2011 22:02:52 -0400
Thread-Topic: [splices] Using Two Separate Devices to Start a Conversation proposal
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Hi All,

Very interesting discussion.

What I had in mind is what Peter described below, but you guys came up with=
 very interesting new ideas that can be used to address this scenario.
The scenario is technically an interesting and challenging one, but I tend =
to agree with Paul that it is not really a use case that will likely be use=
d and maybe we should spend our time addressing other more likely to be use=
d uses case.

Regards,
 Rifaat



________________________________________
From: splices-bounces@ietf.org [splices-bounces@ietf.org] On Behalf Of Paul=
 Kyzivat [pkyzivat@cisco.com]
Sent: Monday, June 06, 2011 7:45 PM
To: splices@ietf.org
Subject: Re: [splices] Using Two Separate Devices to Start a Conversation p=
roposal

I suspect asymmetric rtp *won't* work a lot of the time, for many
reasons, whether we think they are reasonable or not.

Maybe this is just an unreasonable use case. I can't really imagine
wanting to do this. If I have a handset, or headset, with a mic, it
probably has a "speaker" too, and it would seem weird using the mic and
not its speaker too. And when they are separate there is likely to be
problems with echos.

        Thanks,
        Paul

On 6/6/2011 7:27 PM, Peter Musgrave wrote:
> Hi all,
>
> My assumptions were that:
> - Alice's desk phone is doing the media aggregation and creating a
> reINVITE for Bob with a=3Dsendrecv and asymmetric media:
>    * media is sent from Bob to Alice desk phone
>    * media is sent from Alice Computer to Bob
>
> As far as Bob knows this is a dialog only with Alice's deskphone. Bob
> does not have to support SPLICES or do anything special.
>
> Bob must handle asymmetric RTP from Alice.
>
> Ok?
>
> I agree that requiring media relay does become a bad idea when the
> relaying element is e.g a cellphone.
>
> Do we think that having asymmetric RTP will be ok? A lot of
> implementation have worked toward symmetric RTP (and some will only play
> RTP arriving from the place they are sending to! [1 implementation at
> SIPIT28, 3 at SIPIT27 - so maybe not worth worrying about]). Are there
> any issues with ZRTP or use of ICE? [I am not an ICE expert so I can't
> recall if symmetric RTP was baked into ICE, umm "frozen in" to ICE maybe?=
]
>
> Regards,
>
> Peter
>
>
>
>
>
> On Mon, Jun 6, 2011 at 5:45 PM, Alan Johnston <alan.b.johnston@gmail.com
> <mailto:alan.b.johnston@gmail.com>> wrote:
>
>     But isn't the INVITE Join received at Alice's Desk Phone?  If so, the=
n
>     a normal Join will result in Alice's Desk Phone performing mixing of
>     audio streams from Alice's PC, Alice's Desk Phone, and Bob.  I though=
t
>     what was desired (and shown by Rifaat's flow) was that Alice's Desk
>     phone did not mix the media, but instead inserted/spliced the media
>     from Alice's PC into the dialog with Bob, resulting in a re-INVITE to
>     Bob with two m=3D lines.
>
>     - Alan -
>
>     On Mon, Jun 6, 2011 at 4:15 PM, Worley, Dale R (Dale)
>     <dworley@avaya.com <mailto:dworley@avaya.com>> wrote:
>      > No, surely it is an ordinary join.  At the far UA, there are two
>     incoming dialogs, each with one audio m=3D line.  The only oddity is
>     that one dialog is recvonly (at the far UA), and so is never sent
>     media.  The other dialog is sendonly (at the far UA) and so never
>     provides audio to the mixing, but is only sent the result of the
>     mixing.  The effect is that the far UA's input audio is sent only to
>     the second dialog, and the far UA's output audio is driven only by
>     the first dialog.
>      >
>      > Dale
>      >
>     _______________________________________________
>     splices mailing list
>     splices@ietf.org <mailto:splices@ietf.org>
>     https://www.ietf.org/mailman/listinfo/splices
>
>
>
>
> _______________________________________________
> splices mailing list
> splices@ietf.org
> https://www.ietf.org/mailman/listinfo/splices
_______________________________________________
splices mailing list
splices@ietf.org
https://www.ietf.org/mailman/listinfo/splices=

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From: Peter Musgrave <musgravepj@gmail.com>
Date: Tue, 7 Jun 2011 06:19:10 -0400
To: "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>
Cc: "splices@ietf.org" <splices@ietf.org>
Subject: Re: [splices] Using Two Separate Devices to Start a Conversation proposal
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I agree this can be dropped.

Peter

Sent from my iPad

On 2011-06-06, at 10:02 PM, "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.co=
m> wrote:

> Hi All,
>=20
> Very interesting discussion.
>=20
> What I had in mind is what Peter described below, but you guys came up wit=
h very interesting new ideas that can be used to address this scenario.
> The scenario is technically an interesting and challenging one, but I tend=
 to agree with Paul that it is not really a use case that will likely be use=
d and maybe we should spend our time addressing other more likely to be used=
 uses case.
>=20
> Regards,
> Rifaat
>=20
>=20
>=20
> ________________________________________
> From: splices-bounces@ietf.org [splices-bounces@ietf.org] On Behalf Of Pau=
l Kyzivat [pkyzivat@cisco.com]
> Sent: Monday, June 06, 2011 7:45 PM
> To: splices@ietf.org
> Subject: Re: [splices] Using Two Separate Devices to Start a Conversation p=
roposal
>=20
> I suspect asymmetric rtp *won't* work a lot of the time, for many
> reasons, whether we think they are reasonable or not.
>=20
> Maybe this is just an unreasonable use case. I can't really imagine
> wanting to do this. If I have a handset, or headset, with a mic, it
> probably has a "speaker" too, and it would seem weird using the mic and
> not its speaker too. And when they are separate there is likely to be
> problems with echos.
>=20
>        Thanks,
>        Paul
>=20
> On 6/6/2011 7:27 PM, Peter Musgrave wrote:
>> Hi all,
>>=20
>> My assumptions were that:
>> - Alice's desk phone is doing the media aggregation and creating a
>> reINVITE for Bob with a=3Dsendrecv and asymmetric media:
>>   * media is sent from Bob to Alice desk phone
>>   * media is sent from Alice Computer to Bob
>>=20
>> As far as Bob knows this is a dialog only with Alice's deskphone. Bob
>> does not have to support SPLICES or do anything special.
>>=20
>> Bob must handle asymmetric RTP from Alice.
>>=20
>> Ok?
>>=20
>> I agree that requiring media relay does become a bad idea when the
>> relaying element is e.g a cellphone.
>>=20
>> Do we think that having asymmetric RTP will be ok? A lot of
>> implementation have worked toward symmetric RTP (and some will only play
>> RTP arriving from the place they are sending to! [1 implementation at
>> SIPIT28, 3 at SIPIT27 - so maybe not worth worrying about]). Are there
>> any issues with ZRTP or use of ICE? [I am not an ICE expert so I can't
>> recall if symmetric RTP was baked into ICE, umm "frozen in" to ICE maybe?=
]
>>=20
>> Regards,
>>=20
>> Peter
>>=20
>>=20
>>=20
>>=20
>>=20
>> On Mon, Jun 6, 2011 at 5:45 PM, Alan Johnston <alan.b.johnston@gmail.com
>> <mailto:alan.b.johnston@gmail.com>> wrote:
>>=20
>>    But isn't the INVITE Join received at Alice's Desk Phone?  If so, then=

>>    a normal Join will result in Alice's Desk Phone performing mixing of
>>    audio streams from Alice's PC, Alice's Desk Phone, and Bob.  I thought=

>>    what was desired (and shown by Rifaat's flow) was that Alice's Desk
>>    phone did not mix the media, but instead inserted/spliced the media
>>    from Alice's PC into the dialog with Bob, resulting in a re-INVITE to
>>    Bob with two m=3D lines.
>>=20
>>    - Alan -
>>=20
>>    On Mon, Jun 6, 2011 at 4:15 PM, Worley, Dale R (Dale)
>>    <dworley@avaya.com <mailto:dworley@avaya.com>> wrote:
>>> No, surely it is an ordinary join.  At the far UA, there are two
>>    incoming dialogs, each with one audio m=3D line.  The only oddity is
>>    that one dialog is recvonly (at the far UA), and so is never sent
>>    media.  The other dialog is sendonly (at the far UA) and so never
>>    provides audio to the mixing, but is only sent the result of the
>>    mixing.  The effect is that the far UA's input audio is sent only to
>>    the second dialog, and the far UA's output audio is driven only by
>>    the first dialog.
>>>=20
>>> Dale
>>>=20
>>    _______________________________________________
>>    splices mailing list
>>    splices@ietf.org <mailto:splices@ietf.org>
>>    https://www.ietf.org/mailman/listinfo/splices
>>=20
>>=20
>>=20
>>=20
>> _______________________________________________
>> splices mailing list
>> splices@ietf.org
>> https://www.ietf.org/mailman/listinfo/splices
> _______________________________________________
> splices mailing list
> splices@ietf.org
> https://www.ietf.org/mailman/listinfo/splices
> _______________________________________________
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From dworley@avaya.com  Tue Jun  7 08:56:30 2011
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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>, "splices@ietf.org" <splices@ietf.org>
Date: Tue, 7 Jun 2011 11:56:06 -0400
Thread-Topic: [splices] Initiate an A/V Call Using Two Separate Devices proposal
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Subject: Re: [splices] Initiate an A/V Call Using Two Separate Devices proposal
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These are interesting call flows, and seem to be based on the idea of havin=
g a 3PCC-style aggregation controller in the call.

I think that the INVOKE URNs could be made simpler and more systematic.  Th=
e problem with the ones shown is that they depend on particular media types=
 implicitly triggering the desired behavior; I envision an ever-expanding l=
ist of URNs for specific circumstances.  Instead, how about constructing on=
e URN whose purpose is to tell the target to initiate a call, but to specif=
y one or more other UAs as "aggregation slaves":  <urn:invoke:call:initiate=
;aggregate=3D[my URI];target=3Dsip:bob@whatever>.  Or we may want to add a =
parameter meaning "only accept a particular media type from this aggregatio=
n slave":  <urn:invoke:call:initiate;aggregate=3D[my URI];media=3Dvideo;tar=
get=3Dsip:bob@whatever>.

Both of the given examples seem to have the same pattern:  The device from =
which the user is initiating the call isn't the one that is the aggregation=
 controller.  Shouldn't there be one example where it is and one where it i=
sn't?

Dale

From rifatyu@avaya.com  Fri Jun 10 06:53:11 2011
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From: "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>
To: "Worley, Dale R (Dale)" <dworley@avaya.com>, "splices@ietf.org" <splices@ietf.org>
Date: Fri, 10 Jun 2011 09:52:46 -0400
Thread-Topic: [splices] Initiate an A/V Call Using Two Separate Devices proposal
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Dale,

Can you provide an example use case for these new actions you mentioned bel=
ow?
Are these to replace or in addition to the ones we have been using so far?

For the second part of you comment below, take a look at the following flow=
s.
The INVOKE method approach is flexible enough to allow any party to take th=
e role of the Controller.

Regards,
 Rifaat


   Alice                 Alice                Proxy                  Bob
    PC                 Desk Phone                                     =20
    |                     |                     |                     |
    |                     |                     |                     |
Both Alice's devices subscribe to the reg event package, which allows each =
device to=20
discover the capabilities of the other device based on the feature tags pro=
vided by each device.
The Desk Phone knows that the PC supports Video, while the PC knows that th=
e Desk Phone only supports audio.
    |                     |                     |                     |
    |                     | SUBSCRIBE reg       |                     |
    |                     |-------------------->|                     |
    |                     | 200 OK              |                     |
    |                     |<--------------------|                     |
    |                     |                     |                     |
    | SUBSCRIBE reg       |                     |                     |
    |------------------------------------------>|                     |
    | 200 OK              |                     |                     |
    |<------------------------------------------|                     |
    |                     |                     |                     |
    |                     |                     |                     |
The two devices also subscribe to the dialog of each other.
    |                     |                     |                     |
    |                     |                     |                     |
    | SUBSCRIBE dialog    |                     |                     |
    |-------------------->|                     |                     |
    | 200 OK              |                     |                     |
    |<--------------------|                     |                     |
    |                     |                     |                     |
    | SUBSCRIBE dialog    |                     |                     |
    |<--------------------|                     |                     |
    | 200 OK              |                     |                     |
    |-------------------->|                     |                     |
    |                     |                     |                     |
    |                     |                     |                     |





Alice wants to use her PC to initiate an A/V call to Bob.
The PC instructs the phone to initiate an audio call, but does not provide =
a target,=20
which indicates to the phone to send it to the PC.
When the PC gets the audio offer from the phone it sends an A/V offer to th=
e remote=20
party and when it gets the A/V answer it sends an audio answer to the phone=
.
    |                     |                     |                     |
    |                     |                     |                     |
    | INVOKE Action: urn:invoke:call:initiate;media=3Daudio
    |-------------------->|                     |                     |
    | 200 OK              |                     |                     |
    |<--------------------|                     |                     |
    |                     |                     |                     |
    | INVITE [Audio]      |                     |                     |
    |<--------------------|                     |                     |
    |                     |                     |                     |
    | INVITE [A/V]        |                     |                     |
    |------------------------------------------>|                     |
    |                     |                     | INVITE [A/V]        |
    |                     |                     |-------------------->|
    |                     |                     | 180                 |
    | 180                 |                     |<--------------------|
    |<------------------------------------------|                     |
    | 180                 |                     |                     |
    |-------------------->|                     |                     |
    |                     |                     | 200 OK [A/V]        |
    | 200 OK [A/V]        |                     |<--------------------|
    |<------------------------------------------|                     |
    |                     |                     |                     |
    | 200 OK [Audio]      |                     |                     |
    |-------------------->|                     |                     |
    |                     |                     |                     |
    |                     |                     |                     |
    |<-------------------------dialog1------------------------------->|
    |<----dialog2-------->|                     |                     |
    |                     |                     |                     |
    |                     |<=3D=3D=3D=3D=3D=3Daudio=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D>|
    |<=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3Dvideo=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D>|
    |                     |                     |                     |





Alice wants to use her phone to initiate an A/V call to Bob.
The phone knows it cannot handle video, but it know that the PC can.=20


The phone sends an INVOKE request to instruct the PC to initiate a vidoe ca=
ll.=20
Because the Action does not have a target parameter, this indicates to the =
PC to send the INVITE to the phone.
    |                     |                     |                     |
    | INVOKE Action: urn:invoke:call:initiate;media=3Dvideo
    |<--------------------|                     |                     |
    | 200 OK              |                     |                     |
    |-------------------->|                     |                     |
    |                     |                     |                     |
    | INVITE [Video]      |                     |                     |
    |-------------------->|                     |                     |
    | 100                 |                     |                     |
    |<--------------------|                     |                     |
    |                     | INVITE [A/V]        |                     |
    |                     |-------------------->|                     |
    |                     |                     | INVITE [A/V]        |
    |                     |                     |-------------------->|
    |                     |                     | 180                 |
    |                     | 180                 |<--------------------|
    |                     |<--------------------|                     |
    | 180                 |                     |                     |
    |<--------------------|                     | 200 OK [A/V]        |
    |                     | 200 OK [A/V]        |<--------------------|
    |                     |<--------------------|                     |
    | ACK [Video]         |                     |                     |
    |<--------------------|                     |                     |
    |                     | ACK                 |                     |
    |                     |-------------------->|                     |
    |                     |                     | ACK                 |
    |                     |                     |-------------------->|
    |                     |                     |                     |
    |<----dialog2-------->|<---dialog1------------------------------->|
    |                     |                     |                     |
    |                     |<=3D=3D=3D=3D=3D=3Daudio=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D>|
    |<=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3Dvideo=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D>|
    |                     |                     |                     |



> -----Original Message-----
> From: Worley, Dale R (Dale)
> Sent: Tuesday, June 07, 2011 11:56 AM
> To: Shekh-Yusef, Rifaat (Rifaat); splices@ietf.org
> Subject: RE: [splices] Initiate an A/V Call Using Two Separate Devices
> proposal
>=20
> These are interesting call flows, and seem to be based on the idea of hav=
ing a
> 3PCC-style aggregation controller in the call.
>=20
> I think that the INVOKE URNs could be made simpler and more systematic.  =
The
> problem with the ones shown is that they depend on particular media types
> implicitly triggering the desired behavior; I envision an ever-expanding =
list
> of URNs for specific circumstances.  Instead, how about constructing one =
URN
> whose purpose is to tell the target to initiate a call, but to specify on=
e or
> more other UAs as "aggregation slaves":
> <urn:invoke:call:initiate;aggregate=3D[my URI];target=3Dsip:bob@whatever>=
.  Or we
> may want to add a parameter meaning "only accept a particular media type =
from
> this aggregation slave":  <urn:invoke:call:initiate;aggregate=3D[my
> URI];media=3Dvideo;target=3Dsip:bob@whatever>.
>=20
> Both of the given examples seem to have the same pattern:  The device fro=
m
> which the user is initiating the call isn't the one that is the aggregati=
on
> controller.  Shouldn't there be one example where it is and one where it
> isn't?
>=20
> Dale

From rifatyu@avaya.com  Fri Jun 10 12:08:37 2011
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From: "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>
To: "splices@ietf.org" <splices@ietf.org>
Date: Fri, 10 Jun 2011 15:08:22 -0400
Thread-Topic: SIP INVOKE method v1
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Hi,

We have just submitted a new version of the SIP INVOKE method draft that do=
es not include implicit subscription.
https://datatracker.ietf.org/doc/draft-yusef-splices-invoke/

We would appreciate it if you review the document and provide us with your =
feedback.

Thanks,
Rifaat


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s.org/package/2006/digital-signature" xmlns:mver=3D"http://schemas.openxmlf=
ormats.org/markup-compatibility/2006" xmlns:m=3D"http://schemas.microsoft.c=
om/office/2004/12/omml" xmlns:mrels=3D"http://schemas.openxmlformats.org/pa=
ckage/2006/relationships" xmlns:spwp=3D"http://microsoft.com/sharepoint/web=
partpages" xmlns:ex12t=3D"http://schemas.microsoft.com/exchange/services/20=
06/types" xmlns:ex12m=3D"http://schemas.microsoft.com/exchange/services/200=
6/messages" xmlns:pptsl=3D"http://schemas.microsoft.com/sharepoint/soap/Sli=
deLibrary/" xmlns:spsl=3D"http://microsoft.com/webservices/SharePointPortal=
Server/PublishedLinksService" xmlns:Z=3D"urn:schemas-microsoft-com:" xmlns:=
st=3D"&#1;" xmlns=3D"http://www.w3.org/TR/REC-html40"><head><meta http-equi=
v=3DContent-Type content=3D"text/html; charset=3Dus-ascii"><meta name=3DGen=
erator content=3D"Microsoft Word 12 (filtered medium)"><style><!--
/* Font Definitions */
@font-face
	{font-family:"Cambria Math";
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	{margin:0in;
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	font-size:11.0pt;
	font-family:"Calibri","sans-serif";}
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	{mso-style-priority:99;
	color:blue;
	text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
	{mso-style-priority:99;
	color:purple;
	text-decoration:underline;}
span.EmailStyle17
	{mso-style-type:personal-compose;
	font-family:"Calibri","sans-serif";
	color:windowtext;}
.MsoChpDefault
	{mso-style-type:export-only;}
@page WordSection1
	{size:8.5in 11.0in;
	margin:1.0in 1.0in 1.0in 1.0in;}
div.WordSection1
	{page:WordSection1;}
--></style><!--[if gte mso 9]><xml>
<o:shapedefaults v:ext=3D"edit" spidmax=3D"1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext=3D"edit">
<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue vli=
nk=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal>Hi,<o:p></o:p></=
p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>We have ju=
st submitted a new version of the SIP INVOKE method draft that does not inc=
lude implicit subscription.<o:p></o:p></p><p class=3DMsoNormal><a href=3D"h=
ttps://datatracker.ietf.org/doc/draft-yusef-splices-invoke/">https://datatr=
acker.ietf.org/doc/draft-yusef-splices-invoke/</a><o:p></o:p></p><p class=
=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>We would appreciate =
it if you review the document and provide us with your feedback.<o:p></o:p>=
</p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Thanks,<=
o:p></o:p></p><p class=3DMsoNormal> Rifaat<o:p></o:p></p><p class=3DMsoNorm=
al><o:p>&nbsp;</o:p></p></div></body></html>=

--_000_6369CB70BFD88942B9705AC1E639A33822CD81384CDCUS1MBEX4glo_--

From pkyzivat@cisco.com  Fri Jun 10 12:45:41 2011
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Date: Fri, 10 Jun 2011 15:45:37 -0400
From: Paul Kyzivat <pkyzivat@cisco.com>
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Subject: Re: [splices] SIP INVOKE method v1
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This is getting better with age!

Splitting off the subscription is good.

An area that still needs work is in the semantics of the urns.
Some of them are more clear than others. Some are "limited" in ways that 
might not matter to some but that could prove a problem in the end.
Specifically:

     o  Answer call       - urn:invoke:call:answer
     o  Terminate call    - urn:invoke:call:terminate
     o  Decline call      - urn:invoke:call:decline
     o  Ignore call       - urn:invoke:call:ignore

The above need some elaboration, but I think its just work - probably 
anybody who does it will end up with an equivalent result.

     o  Send call to VM   - urn:invoke:call:sendvm

VM is a "feature", and we haven't standardized features much, except for 
bliss. How does one know if the recipient understands VM and what to do 
to send a call there? What if there are multiple VM options (e.g. what 
to say, etc.) And what if the call isn't a voice call. (Maybe VM now 
means voice/video mail. But the "call" could be IM.)

     o  Hold call         - urn:invoke:call:hold
     o  Unhold call       - urn:invoke:call:unhold
     o  Mute call         - urn:invoke:call:mute
     o  Unmute call       - urn:invoke:call:unmute

Similarly these are all "features" that we all have some expectations 
about, but that are not rigorously defined. (But perhaps Mute/Unmute 
could be formalized by using the 'transducer' option. (Transducer=none?)

My concern with all these "features" is that we are being put into a box 
where we must either:
- define them with sufficient rigor
- fall back on "do what I mean", which has proved to be a bad idea,
   because not everybody has the same notion of what they mean.

     o  Conference        - urn:invoke:conference:add
                          - urn:invoke:conference:remove

This one may be clearer due to the existence of xcon specs.

Regarding action parameters:

     o  media=audio|video|audvid

I'm troubled by audvid. I would rather see "audio,video". Specifically, 
what is the namespace here, and what does specifying it mean? I think 
the namespace could be: one or more values allowable as the <media> 
parameter in the m-line in SDP.

Defining exactly what it means when present is a little harder. When 
answering a call that has an offer, I guess it could mean: refuse any 
m-lines with media type that doesn't match one of these, and accept as 
many of the remaining as you can.

For answering an offerless invite, I guess it could mean: offer whatever 
media you are capable of offering that match one of these, but don't 
offer anything that doesn't match.

Is there likely to be a need to get more specific? E.g. about what 
codecs to use? (Hopefully not - that could get very messy.)

     o  transducer=speaker|headset

My deskphone has three possibilities: speaker|headset|handset.
And I'm far from certain this is exhaustive. (The clue wg is defining a 
much more complex environment.) This at least needs to be extensible. We 
may need some sort of extended model of a UA that includes this sort of 
stuff.

     o  target=<AOR>
     o  direction=recvonly|sendonly|sendrecv
     o  abort

None of this is intended as objection to the work so far. Its just a 
start at identifying what else there is to do. The whole notion of 
naming features to invoke causes me the most concern.

	Thanks,
	Paul

On 6/10/2011 3:08 PM, Shekh-Yusef, Rifaat (Rifaat) wrote:
> Hi,
>
> We have just submitted a new version of the SIP INVOKE method draft that
> does not include implicit subscription.
>
> https://datatracker.ietf.org/doc/draft-yusef-splices-invoke/
>
> We would appreciate it if you review the document and provide us with
> your feedback.
>
> Thanks,
>
> Rifaat
>
>
>
> _______________________________________________
> splices mailing list
> splices@ietf.org
> https://www.ietf.org/mailman/listinfo/splices

From andrew.hutton@siemens-enterprise.com  Mon Jun 13 07:04:08 2011
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, "splices@ietf.org" <splices@ietf.org>
Date: Mon, 13 Jun 2011 16:04:03 +0200
Thread-Topic: [splices] SIP INVOKE method v1
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Hi All,

Whilst I still question whether the mechanism described in this draft actua=
lly falls within the SPLICES WG charter as it defines new mechanism I have =
a few comments.

I tend to agree with Paul's comments below but would go further to say that=
 even some of the supposedly simply actions such as "decline" & "ignore" ar=
e also features which will be difficult to agree the semantics for. All of =
the actions will need to be specified rigorously a quick look at ECMA-269 w=
hich specifies these kind of services for CSTA will give you an idea of wha=
t would need to be done. I don't believe that even in the CSTA world which =
defines a very large number of services the services "ignore" or "decline" =
were specified.

With regard to the discussion on headers vs body the draft states.

"There has been some discussion on the list on the question of header field=
 vs message body.  This is not the real issue, as either could conceivably =
be used.  Instead, the main issue is what is being invoked:  is it a named =
feature/action/event, or is it a script".

I must have missed the discussion about scripting but I don't think this is=
 the issue. The issue is I believe extensibility which any mechanism for in=
voking services will need and this is the reason why an XML body would be b=
etter than using SIP headers. =20

Regards
Andy


> -----Original Message-----
> From: splices-bounces@ietf.org [mailto:splices-bounces@ietf.org] On
> Behalf Of Paul Kyzivat
> Sent: 10 June 2011 20:46
> To: splices@ietf.org
> Subject: Re: [splices] SIP INVOKE method v1
>=20
> This is getting better with age!
>=20
> Splitting off the subscription is good.
>=20
> An area that still needs work is in the semantics of the urns.
> Some of them are more clear than others. Some are "limited" in ways
> that
> might not matter to some but that could prove a problem in the end.
> Specifically:
>=20
>      o  Answer call       - urn:invoke:call:answer
>      o  Terminate call    - urn:invoke:call:terminate
>      o  Decline call      - urn:invoke:call:decline
>      o  Ignore call       - urn:invoke:call:ignore
>=20
> The above need some elaboration, but I think its just work - probably
> anybody who does it will end up with an equivalent result.
>=20
>      o  Send call to VM   - urn:invoke:call:sendvm
>=20
> VM is a "feature", and we haven't standardized features much, except
> for
> bliss. How does one know if the recipient understands VM and what to do
> to send a call there? What if there are multiple VM options (e.g. what
> to say, etc.) And what if the call isn't a voice call. (Maybe VM now
> means voice/video mail. But the "call" could be IM.)
>=20
>      o  Hold call         - urn:invoke:call:hold
>      o  Unhold call       - urn:invoke:call:unhold
>      o  Mute call         - urn:invoke:call:mute
>      o  Unmute call       - urn:invoke:call:unmute
>=20
> Similarly these are all "features" that we all have some expectations
> about, but that are not rigorously defined. (But perhaps Mute/Unmute
> could be formalized by using the 'transducer' option.
> (Transducer=3Dnone?)
>=20
> My concern with all these "features" is that we are being put into a
> box
> where we must either:
> - define them with sufficient rigor
> - fall back on "do what I mean", which has proved to be a bad idea,
>    because not everybody has the same notion of what they mean.
>=20
>      o  Conference        - urn:invoke:conference:add
>                           - urn:invoke:conference:remove
>=20
> This one may be clearer due to the existence of xcon specs.
>=20
> Regarding action parameters:
>=20
>      o  media=3Daudio|video|audvid
>=20
> I'm troubled by audvid. I would rather see "audio,video". Specifically,
> what is the namespace here, and what does specifying it mean? I think
> the namespace could be: one or more values allowable as the <media>
> parameter in the m-line in SDP.
>=20
> Defining exactly what it means when present is a little harder. When
> answering a call that has an offer, I guess it could mean: refuse any
> m-lines with media type that doesn't match one of these, and accept as
> many of the remaining as you can.
>=20
> For answering an offerless invite, I guess it could mean: offer
> whatever
> media you are capable of offering that match one of these, but don't
> offer anything that doesn't match.
>=20
> Is there likely to be a need to get more specific? E.g. about what
> codecs to use? (Hopefully not - that could get very messy.)
>=20
>      o  transducer=3Dspeaker|headset
>=20
> My deskphone has three possibilities: speaker|headset|handset.
> And I'm far from certain this is exhaustive. (The clue wg is defining a
> much more complex environment.) This at least needs to be extensible.
> We
> may need some sort of extended model of a UA that includes this sort of
> stuff.
>=20
>      o  target=3D<AOR>
>      o  direction=3Drecvonly|sendonly|sendrecv
>      o  abort
>=20
> None of this is intended as objection to the work so far. Its just a
> start at identifying what else there is to do. The whole notion of
> naming features to invoke causes me the most concern.
>=20
> 	Thanks,
> 	Paul
>=20
> On 6/10/2011 3:08 PM, Shekh-Yusef, Rifaat (Rifaat) wrote:
> > Hi,
> >
> > We have just submitted a new version of the SIP INVOKE method draft
> that
> > does not include implicit subscription.
> >
> > https://datatracker.ietf.org/doc/draft-yusef-splices-invoke/
> >
> > We would appreciate it if you review the document and provide us with
> > your feedback.
> >
> > Thanks,
> >
> > Rifaat
> >
> >
> >
> > _______________________________________________
> > splices mailing list
> > splices@ietf.org
> > https://www.ietf.org/mailman/listinfo/splices
> _______________________________________________
> splices mailing list
> splices@ietf.org
> https://www.ietf.org/mailman/listinfo/splices

From rifatyu@avaya.com  Tue Jun 14 09:30:38 2011
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From: "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, "splices@ietf.org" <splices@ietf.org>
Date: Tue, 14 Jun 2011 12:30:15 -0400
Thread-Topic: [splices] SIP INVOKE method v1
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Hi Paul,

Thanks for the detailed review and feedback.
I agree with you that this is the next thing that we need to describe in de=
tails.
I will be working on providing more details to explain our thinking around =
the various actions and parameters.

Regards,
 Rifaat



> -----Original Message-----
> From: splices-bounces@ietf.org [mailto:splices-bounces@ietf.org] On Behal=
f Of
> Paul Kyzivat
> Sent: Friday, June 10, 2011 3:46 PM
> To: splices@ietf.org
> Subject: Re: [splices] SIP INVOKE method v1
>=20
> This is getting better with age!
>=20
> Splitting off the subscription is good.
>=20
> An area that still needs work is in the semantics of the urns.
> Some of them are more clear than others. Some are "limited" in ways that
> might not matter to some but that could prove a problem in the end.
> Specifically:
>=20
>      o  Answer call       - urn:invoke:call:answer
>      o  Terminate call    - urn:invoke:call:terminate
>      o  Decline call      - urn:invoke:call:decline
>      o  Ignore call       - urn:invoke:call:ignore
>=20
> The above need some elaboration, but I think its just work - probably
> anybody who does it will end up with an equivalent result.
>=20
>      o  Send call to VM   - urn:invoke:call:sendvm
>=20
> VM is a "feature", and we haven't standardized features much, except for
> bliss. How does one know if the recipient understands VM and what to do
> to send a call there? What if there are multiple VM options (e.g. what
> to say, etc.) And what if the call isn't a voice call. (Maybe VM now
> means voice/video mail. But the "call" could be IM.)
>=20
>      o  Hold call         - urn:invoke:call:hold
>      o  Unhold call       - urn:invoke:call:unhold
>      o  Mute call         - urn:invoke:call:mute
>      o  Unmute call       - urn:invoke:call:unmute
>=20
> Similarly these are all "features" that we all have some expectations
> about, but that are not rigorously defined. (But perhaps Mute/Unmute
> could be formalized by using the 'transducer' option. (Transducer=3Dnone?=
)
>=20
> My concern with all these "features" is that we are being put into a box
> where we must either:
> - define them with sufficient rigor
> - fall back on "do what I mean", which has proved to be a bad idea,
>    because not everybody has the same notion of what they mean.
>=20
>      o  Conference        - urn:invoke:conference:add
>                           - urn:invoke:conference:remove
>=20
> This one may be clearer due to the existence of xcon specs.
>=20
> Regarding action parameters:
>=20
>      o  media=3Daudio|video|audvid
>=20
> I'm troubled by audvid. I would rather see "audio,video". Specifically,
> what is the namespace here, and what does specifying it mean? I think
> the namespace could be: one or more values allowable as the <media>
> parameter in the m-line in SDP.
>=20
> Defining exactly what it means when present is a little harder. When
> answering a call that has an offer, I guess it could mean: refuse any
> m-lines with media type that doesn't match one of these, and accept as
> many of the remaining as you can.
>=20
> For answering an offerless invite, I guess it could mean: offer whatever
> media you are capable of offering that match one of these, but don't
> offer anything that doesn't match.
>=20
> Is there likely to be a need to get more specific? E.g. about what
> codecs to use? (Hopefully not - that could get very messy.)
>=20
>      o  transducer=3Dspeaker|headset
>=20
> My deskphone has three possibilities: speaker|headset|handset.
> And I'm far from certain this is exhaustive. (The clue wg is defining a
> much more complex environment.) This at least needs to be extensible. We
> may need some sort of extended model of a UA that includes this sort of
> stuff.
>=20
>      o  target=3D<AOR>
>      o  direction=3Drecvonly|sendonly|sendrecv
>      o  abort
>=20
> None of this is intended as objection to the work so far. Its just a
> start at identifying what else there is to do. The whole notion of
> naming features to invoke causes me the most concern.
>=20
> 	Thanks,
> 	Paul
>=20
> On 6/10/2011 3:08 PM, Shekh-Yusef, Rifaat (Rifaat) wrote:
> > Hi,
> >
> > We have just submitted a new version of the SIP INVOKE method draft tha=
t
> > does not include implicit subscription.
> >
> > https://datatracker.ietf.org/doc/draft-yusef-splices-invoke/
> >
> > We would appreciate it if you review the document and provide us with
> > your feedback.
> >
> > Thanks,
> >
> > Rifaat
> >
> >
> >
> > _______________________________________________
> > splices mailing list
> > splices@ietf.org
> > https://www.ietf.org/mailman/listinfo/splices
> _______________________________________________
> splices mailing list
> splices@ietf.org
> https://www.ietf.org/mailman/listinfo/splices

From rifatyu@avaya.com  Tue Jun 14 09:32:16 2011
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From: "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>
To: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>, Paul Kyzivat <pkyzivat@cisco.com>, "splices@ietf.org" <splices@ietf.org>
Date: Tue, 14 Jun 2011 12:32:13 -0400
Thread-Topic: [splices] SIP INVOKE method v1
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Andy,

> The issue is I believe extensibility which any mechanism for
> invoking services will need and this is the reason why an XML body would =
be
> better than using SIP headers.

What makes you think that the current proposal is not extensible?

Regards,
 Rifaat


> -----Original Message-----
> From: splices-bounces@ietf.org [mailto:splices-bounces@ietf.org] On Behal=
f Of
> Hutton, Andrew
> Sent: Monday, June 13, 2011 10:04 AM
> To: Paul Kyzivat; splices@ietf.org
> Subject: Re: [splices] SIP INVOKE method v1
>=20
> Hi All,
>=20
> Whilst I still question whether the mechanism described in this draft act=
ually
> falls within the SPLICES WG charter as it defines new mechanism I have a =
few
> comments.
>=20
> I tend to agree with Paul's comments below but would go further to say th=
at
> even some of the supposedly simply actions such as "decline" & "ignore" a=
re
> also features which will be difficult to agree the semantics for. All of =
the
> actions will need to be specified rigorously a quick look at ECMA-269 whi=
ch
> specifies these kind of services for CSTA will give you an idea of what w=
ould
> need to be done. I don't believe that even in the CSTA world which define=
s a
> very large number of services the services "ignore" or "decline" were
> specified.
>=20
> With regard to the discussion on headers vs body the draft states.
>=20
> "There has been some discussion on the list on the question of header fie=
ld vs
> message body.  This is not the real issue, as either could conceivably be
> used.  Instead, the main issue is what is being invoked:  is it a named
> feature/action/event, or is it a script".
>=20
> I must have missed the discussion about scripting but I don't think this =
is
> the issue. The issue is I believe extensibility which any mechanism for
> invoking services will need and this is the reason why an XML body would =
be
> better than using SIP headers.
>=20
> Regards
> Andy
>=20
>=20
> > -----Original Message-----
> > From: splices-bounces@ietf.org [mailto:splices-bounces@ietf.org] On
> > Behalf Of Paul Kyzivat
> > Sent: 10 June 2011 20:46
> > To: splices@ietf.org
> > Subject: Re: [splices] SIP INVOKE method v1
> >
> > This is getting better with age!
> >
> > Splitting off the subscription is good.
> >
> > An area that still needs work is in the semantics of the urns.
> > Some of them are more clear than others. Some are "limited" in ways
> > that
> > might not matter to some but that could prove a problem in the end.
> > Specifically:
> >
> >      o  Answer call       - urn:invoke:call:answer
> >      o  Terminate call    - urn:invoke:call:terminate
> >      o  Decline call      - urn:invoke:call:decline
> >      o  Ignore call       - urn:invoke:call:ignore
> >
> > The above need some elaboration, but I think its just work - probably
> > anybody who does it will end up with an equivalent result.
> >
> >      o  Send call to VM   - urn:invoke:call:sendvm
> >
> > VM is a "feature", and we haven't standardized features much, except
> > for
> > bliss. How does one know if the recipient understands VM and what to do
> > to send a call there? What if there are multiple VM options (e.g. what
> > to say, etc.) And what if the call isn't a voice call. (Maybe VM now
> > means voice/video mail. But the "call" could be IM.)
> >
> >      o  Hold call         - urn:invoke:call:hold
> >      o  Unhold call       - urn:invoke:call:unhold
> >      o  Mute call         - urn:invoke:call:mute
> >      o  Unmute call       - urn:invoke:call:unmute
> >
> > Similarly these are all "features" that we all have some expectations
> > about, but that are not rigorously defined. (But perhaps Mute/Unmute
> > could be formalized by using the 'transducer' option.
> > (Transducer=3Dnone?)
> >
> > My concern with all these "features" is that we are being put into a
> > box
> > where we must either:
> > - define them with sufficient rigor
> > - fall back on "do what I mean", which has proved to be a bad idea,
> >    because not everybody has the same notion of what they mean.
> >
> >      o  Conference        - urn:invoke:conference:add
> >                           - urn:invoke:conference:remove
> >
> > This one may be clearer due to the existence of xcon specs.
> >
> > Regarding action parameters:
> >
> >      o  media=3Daudio|video|audvid
> >
> > I'm troubled by audvid. I would rather see "audio,video". Specifically,
> > what is the namespace here, and what does specifying it mean? I think
> > the namespace could be: one or more values allowable as the <media>
> > parameter in the m-line in SDP.
> >
> > Defining exactly what it means when present is a little harder. When
> > answering a call that has an offer, I guess it could mean: refuse any
> > m-lines with media type that doesn't match one of these, and accept as
> > many of the remaining as you can.
> >
> > For answering an offerless invite, I guess it could mean: offer
> > whatever
> > media you are capable of offering that match one of these, but don't
> > offer anything that doesn't match.
> >
> > Is there likely to be a need to get more specific? E.g. about what
> > codecs to use? (Hopefully not - that could get very messy.)
> >
> >      o  transducer=3Dspeaker|headset
> >
> > My deskphone has three possibilities: speaker|headset|handset.
> > And I'm far from certain this is exhaustive. (The clue wg is defining a
> > much more complex environment.) This at least needs to be extensible.
> > We
> > may need some sort of extended model of a UA that includes this sort of
> > stuff.
> >
> >      o  target=3D<AOR>
> >      o  direction=3Drecvonly|sendonly|sendrecv
> >      o  abort
> >
> > None of this is intended as objection to the work so far. Its just a
> > start at identifying what else there is to do. The whole notion of
> > naming features to invoke causes me the most concern.
> >
> > 	Thanks,
> > 	Paul
> >
> > On 6/10/2011 3:08 PM, Shekh-Yusef, Rifaat (Rifaat) wrote:
> > > Hi,
> > >
> > > We have just submitted a new version of the SIP INVOKE method draft
> > that
> > > does not include implicit subscription.
> > >
> > > https://datatracker.ietf.org/doc/draft-yusef-splices-invoke/
> > >
> > > We would appreciate it if you review the document and provide us with
> > > your feedback.
> > >
> > > Thanks,
> > >
> > > Rifaat
> > >
> > >
> > >
> > > _______________________________________________
> > > splices mailing list
> > > splices@ietf.org
> > > https://www.ietf.org/mailman/listinfo/splices
> > _______________________________________________
> > splices mailing list
> > splices@ietf.org
> > https://www.ietf.org/mailman/listinfo/splices
> _______________________________________________
> splices mailing list
> splices@ietf.org
> https://www.ietf.org/mailman/listinfo/splices

From andrew.hutton@siemens-enterprise.com  Tue Jun 14 10:18:33 2011
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>, Paul Kyzivat <pkyzivat@cisco.com>, "splices@ietf.org" <splices@ietf.org>
Date: Tue, 14 Jun 2011 19:18:28 +0200
Thread-Topic: [splices] SIP INVOKE method v1
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Hi Rifaat,

I am not saying that the current proposal is not extensible but I believe t=
hat an XML body would be a much better solution to the extensibility proble=
m that using a SIP header.

The list of potential actions and associated parameters is huge and for thi=
s kind of problem I believe that an XML body would be a much better solutio=
n. I guess I much prefer XML to BNF.

There is also an architectural issue as the actions don't relate directly t=
o the management of the SIP session and therefore most likely need to passe=
d to the application layer in the SIP UA to handle. It is much cleaner arch=
itecturally if the body can simply be removed and passed to the responsible=
 component in the UA rather than the SIP component having to extract the in=
formation from SIP headers.

Regards
Andy




> -----Original Message-----
> From: Shekh-Yusef, Rifaat (Rifaat) [mailto:rifatyu@avaya.com]
> Sent: 14 June 2011 17:32
> To: Hutton, Andrew; Paul Kyzivat; splices@ietf.org
> Subject: RE: [splices] SIP INVOKE method v1
>=20
> Andy,
>=20
> > The issue is I believe extensibility which any mechanism for
> > invoking services will need and this is the reason why an XML body
> would be
> > better than using SIP headers.
>=20
> What makes you think that the current proposal is not extensible?
>=20
> Regards,
>  Rifaat
>=20
>=20
> > -----Original Message-----
> > From: splices-bounces@ietf.org [mailto:splices-bounces@ietf.org] On
> Behalf Of
> > Hutton, Andrew
> > Sent: Monday, June 13, 2011 10:04 AM
> > To: Paul Kyzivat; splices@ietf.org
> > Subject: Re: [splices] SIP INVOKE method v1
> >
> > Hi All,
> >
> > Whilst I still question whether the mechanism described in this draft
> actually
> > falls within the SPLICES WG charter as it defines new mechanism I
> have a few
> > comments.
> >
> > I tend to agree with Paul's comments below but would go further to
> say that
> > even some of the supposedly simply actions such as "decline" &
> "ignore" are
> > also features which will be difficult to agree the semantics for. All
> of the
> > actions will need to be specified rigorously a quick look at ECMA-269
> which
> > specifies these kind of services for CSTA will give you an idea of
> what would
> > need to be done. I don't believe that even in the CSTA world which
> defines a
> > very large number of services the services "ignore" or "decline" were
> > specified.
> >
> > With regard to the discussion on headers vs body the draft states.
> >
> > "There has been some discussion on the list on the question of header
> field vs
> > message body.  This is not the real issue, as either could
> conceivably be
> > used.  Instead, the main issue is what is being invoked:  is it a
> named
> > feature/action/event, or is it a script".
> >
> > I must have missed the discussion about scripting but I don't think
> this is
> > the issue. The issue is I believe extensibility which any mechanism
> for
> > invoking services will need and this is the reason why an XML body
> would be
> > better than using SIP headers.
> >
> > Regards
> > Andy
> >
> >
> > > -----Original Message-----
> > > From: splices-bounces@ietf.org [mailto:splices-bounces@ietf.org] On
> > > Behalf Of Paul Kyzivat
> > > Sent: 10 June 2011 20:46
> > > To: splices@ietf.org
> > > Subject: Re: [splices] SIP INVOKE method v1
> > >
> > > This is getting better with age!
> > >
> > > Splitting off the subscription is good.
> > >
> > > An area that still needs work is in the semantics of the urns.
> > > Some of them are more clear than others. Some are "limited" in ways
> > > that
> > > might not matter to some but that could prove a problem in the end.
> > > Specifically:
> > >
> > >      o  Answer call       - urn:invoke:call:answer
> > >      o  Terminate call    - urn:invoke:call:terminate
> > >      o  Decline call      - urn:invoke:call:decline
> > >      o  Ignore call       - urn:invoke:call:ignore
> > >
> > > The above need some elaboration, but I think its just work -
> probably
> > > anybody who does it will end up with an equivalent result.
> > >
> > >      o  Send call to VM   - urn:invoke:call:sendvm
> > >
> > > VM is a "feature", and we haven't standardized features much,
> except
> > > for
> > > bliss. How does one know if the recipient understands VM and what
> to do
> > > to send a call there? What if there are multiple VM options (e.g.
> what
> > > to say, etc.) And what if the call isn't a voice call. (Maybe VM
> now
> > > means voice/video mail. But the "call" could be IM.)
> > >
> > >      o  Hold call         - urn:invoke:call:hold
> > >      o  Unhold call       - urn:invoke:call:unhold
> > >      o  Mute call         - urn:invoke:call:mute
> > >      o  Unmute call       - urn:invoke:call:unmute
> > >
> > > Similarly these are all "features" that we all have some
> expectations
> > > about, but that are not rigorously defined. (But perhaps
> Mute/Unmute
> > > could be formalized by using the 'transducer' option.
> > > (Transducer=3Dnone?)
> > >
> > > My concern with all these "features" is that we are being put into
> a
> > > box
> > > where we must either:
> > > - define them with sufficient rigor
> > > - fall back on "do what I mean", which has proved to be a bad idea,
> > >    because not everybody has the same notion of what they mean.
> > >
> > >      o  Conference        - urn:invoke:conference:add
> > >                           - urn:invoke:conference:remove
> > >
> > > This one may be clearer due to the existence of xcon specs.
> > >
> > > Regarding action parameters:
> > >
> > >      o  media=3Daudio|video|audvid
> > >
> > > I'm troubled by audvid. I would rather see "audio,video".
> Specifically,
> > > what is the namespace here, and what does specifying it mean? I
> think
> > > the namespace could be: one or more values allowable as the <media>
> > > parameter in the m-line in SDP.
> > >
> > > Defining exactly what it means when present is a little harder.
> When
> > > answering a call that has an offer, I guess it could mean: refuse
> any
> > > m-lines with media type that doesn't match one of these, and accept
> as
> > > many of the remaining as you can.
> > >
> > > For answering an offerless invite, I guess it could mean: offer
> > > whatever
> > > media you are capable of offering that match one of these, but
> don't
> > > offer anything that doesn't match.
> > >
> > > Is there likely to be a need to get more specific? E.g. about what
> > > codecs to use? (Hopefully not - that could get very messy.)
> > >
> > >      o  transducer=3Dspeaker|headset
> > >
> > > My deskphone has three possibilities: speaker|headset|handset.
> > > And I'm far from certain this is exhaustive. (The clue wg is
> defining a
> > > much more complex environment.) This at least needs to be
> extensible.
> > > We
> > > may need some sort of extended model of a UA that includes this
> sort of
> > > stuff.
> > >
> > >      o  target=3D<AOR>
> > >      o  direction=3Drecvonly|sendonly|sendrecv
> > >      o  abort
> > >
> > > None of this is intended as objection to the work so far. Its just
> a
> > > start at identifying what else there is to do. The whole notion of
> > > naming features to invoke causes me the most concern.
> > >
> > > 	Thanks,
> > > 	Paul
> > >
> > > On 6/10/2011 3:08 PM, Shekh-Yusef, Rifaat (Rifaat) wrote:
> > > > Hi,
> > > >
> > > > We have just submitted a new version of the SIP INVOKE method
> draft
> > > that
> > > > does not include implicit subscription.
> > > >
> > > > https://datatracker.ietf.org/doc/draft-yusef-splices-invoke/
> > > >
> > > > We would appreciate it if you review the document and provide us
> with
> > > > your feedback.
> > > >
> > > > Thanks,
> > > >
> > > > Rifaat
> > > >
> > > >
> > > >
> > > > _______________________________________________
> > > > splices mailing list
> > > > splices@ietf.org
> > > > https://www.ietf.org/mailman/listinfo/splices
> > > _______________________________________________
> > > splices mailing list
> > > splices@ietf.org
> > > https://www.ietf.org/mailman/listinfo/splices
> > _______________________________________________
> > splices mailing list
> > splices@ietf.org
> > https://www.ietf.org/mailman/listinfo/splices

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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>, "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>, Paul Kyzivat <pkyzivat@cisco.com>, "splices@ietf.org" <splices@ietf.org>
Date: Tue, 14 Jun 2011 13:32:59 -0400
Thread-Topic: [splices] SIP INVOKE method v1
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> From: Hutton, Andrew [andrew.hutton@siemens-enterprise.com]
>=20
> The list of potential actions and associated parameters is huge and
> for this kind of problem I believe that an XML body would be a much
> better solution. I guess I much prefer XML to BNF.

I'd feel much more comfortable if we settled on either:

1) a small number of actions

2) to use an appropriate subset of an existing CTI standard

In the latter case, could someone provide a pointer to the standard?

Essentially, INVOKE is to provide an interface to a command set, and
we need to have some idea what the outer limits of the command set
will be.  Only then can we choose a good way to represent the commands
within a SIP request.

Dale

From rifatyu@avaya.com  Tue Jun 14 10:42:13 2011
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To: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>, Paul Kyzivat <pkyzivat@cisco.com>, "splices@ietf.org" <splices@ietf.org>
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Andy,

> The list of potential actions and associated parameters is huge and for t=
his
> kind of problem I believe that an XML body would be a much better solutio=
n. I
> guess I much prefer XML to BNF.
>=20
This actually calls for a header rather than an XML body, because the idea =
is to control what get standardized and what not and clearly define the act=
ions that get standardized. With an XML body you actually create a tunnel o=
ver SIP and you lose control over what get defined.

Regards,
 Rifaat


> -----Original Message-----
> From: Hutton, Andrew [mailto:andrew.hutton@siemens-enterprise.com]
> Sent: Tuesday, June 14, 2011 1:18 PM
> To: Shekh-Yusef, Rifaat (Rifaat); Paul Kyzivat; splices@ietf.org
> Subject: RE: [splices] SIP INVOKE method v1
>=20
> Hi Rifaat,
>=20
> I am not saying that the current proposal is not extensible but I believe=
 that
> an XML body would be a much better solution to the extensibility problem =
that
> using a SIP header.
>=20
> The list of potential actions and associated parameters is huge and for t=
his
> kind of problem I believe that an XML body would be a much better solutio=
n. I
> guess I much prefer XML to BNF.
>=20
> There is also an architectural issue as the actions don't relate directly=
 to
> the management of the SIP session and therefore most likely need to passe=
d to
> the application layer in the SIP UA to handle. It is much cleaner
> architecturally if the body can simply be removed and passed to the
> responsible component in the UA rather than the SIP component having to
> extract the information from SIP headers.
>=20
> Regards
> Andy
>=20
>=20
>=20
>=20
> > -----Original Message-----
> > From: Shekh-Yusef, Rifaat (Rifaat) [mailto:rifatyu@avaya.com]
> > Sent: 14 June 2011 17:32
> > To: Hutton, Andrew; Paul Kyzivat; splices@ietf.org
> > Subject: RE: [splices] SIP INVOKE method v1
> >
> > Andy,
> >
> > > The issue is I believe extensibility which any mechanism for
> > > invoking services will need and this is the reason why an XML body
> > would be
> > > better than using SIP headers.
> >
> > What makes you think that the current proposal is not extensible?
> >
> > Regards,
> >  Rifaat
> >
> >
> > > -----Original Message-----
> > > From: splices-bounces@ietf.org [mailto:splices-bounces@ietf.org] On
> > Behalf Of
> > > Hutton, Andrew
> > > Sent: Monday, June 13, 2011 10:04 AM
> > > To: Paul Kyzivat; splices@ietf.org
> > > Subject: Re: [splices] SIP INVOKE method v1
> > >
> > > Hi All,
> > >
> > > Whilst I still question whether the mechanism described in this draft
> > actually
> > > falls within the SPLICES WG charter as it defines new mechanism I
> > have a few
> > > comments.
> > >
> > > I tend to agree with Paul's comments below but would go further to
> > say that
> > > even some of the supposedly simply actions such as "decline" &
> > "ignore" are
> > > also features which will be difficult to agree the semantics for. All
> > of the
> > > actions will need to be specified rigorously a quick look at ECMA-269
> > which
> > > specifies these kind of services for CSTA will give you an idea of
> > what would
> > > need to be done. I don't believe that even in the CSTA world which
> > defines a
> > > very large number of services the services "ignore" or "decline" were
> > > specified.
> > >
> > > With regard to the discussion on headers vs body the draft states.
> > >
> > > "There has been some discussion on the list on the question of header
> > field vs
> > > message body.  This is not the real issue, as either could
> > conceivably be
> > > used.  Instead, the main issue is what is being invoked:  is it a
> > named
> > > feature/action/event, or is it a script".
> > >
> > > I must have missed the discussion about scripting but I don't think
> > this is
> > > the issue. The issue is I believe extensibility which any mechanism
> > for
> > > invoking services will need and this is the reason why an XML body
> > would be
> > > better than using SIP headers.
> > >
> > > Regards
> > > Andy
> > >
> > >
> > > > -----Original Message-----
> > > > From: splices-bounces@ietf.org [mailto:splices-bounces@ietf.org] On
> > > > Behalf Of Paul Kyzivat
> > > > Sent: 10 June 2011 20:46
> > > > To: splices@ietf.org
> > > > Subject: Re: [splices] SIP INVOKE method v1
> > > >
> > > > This is getting better with age!
> > > >
> > > > Splitting off the subscription is good.
> > > >
> > > > An area that still needs work is in the semantics of the urns.
> > > > Some of them are more clear than others. Some are "limited" in ways
> > > > that
> > > > might not matter to some but that could prove a problem in the end.
> > > > Specifically:
> > > >
> > > >      o  Answer call       - urn:invoke:call:answer
> > > >      o  Terminate call    - urn:invoke:call:terminate
> > > >      o  Decline call      - urn:invoke:call:decline
> > > >      o  Ignore call       - urn:invoke:call:ignore
> > > >
> > > > The above need some elaboration, but I think its just work -
> > probably
> > > > anybody who does it will end up with an equivalent result.
> > > >
> > > >      o  Send call to VM   - urn:invoke:call:sendvm
> > > >
> > > > VM is a "feature", and we haven't standardized features much,
> > except
> > > > for
> > > > bliss. How does one know if the recipient understands VM and what
> > to do
> > > > to send a call there? What if there are multiple VM options (e.g.
> > what
> > > > to say, etc.) And what if the call isn't a voice call. (Maybe VM
> > now
> > > > means voice/video mail. But the "call" could be IM.)
> > > >
> > > >      o  Hold call         - urn:invoke:call:hold
> > > >      o  Unhold call       - urn:invoke:call:unhold
> > > >      o  Mute call         - urn:invoke:call:mute
> > > >      o  Unmute call       - urn:invoke:call:unmute
> > > >
> > > > Similarly these are all "features" that we all have some
> > expectations
> > > > about, but that are not rigorously defined. (But perhaps
> > Mute/Unmute
> > > > could be formalized by using the 'transducer' option.
> > > > (Transducer=3Dnone?)
> > > >
> > > > My concern with all these "features" is that we are being put into
> > a
> > > > box
> > > > where we must either:
> > > > - define them with sufficient rigor
> > > > - fall back on "do what I mean", which has proved to be a bad idea,
> > > >    because not everybody has the same notion of what they mean.
> > > >
> > > >      o  Conference        - urn:invoke:conference:add
> > > >                           - urn:invoke:conference:remove
> > > >
> > > > This one may be clearer due to the existence of xcon specs.
> > > >
> > > > Regarding action parameters:
> > > >
> > > >      o  media=3Daudio|video|audvid
> > > >
> > > > I'm troubled by audvid. I would rather see "audio,video".
> > Specifically,
> > > > what is the namespace here, and what does specifying it mean? I
> > think
> > > > the namespace could be: one or more values allowable as the <media>
> > > > parameter in the m-line in SDP.
> > > >
> > > > Defining exactly what it means when present is a little harder.
> > When
> > > > answering a call that has an offer, I guess it could mean: refuse
> > any
> > > > m-lines with media type that doesn't match one of these, and accept
> > as
> > > > many of the remaining as you can.
> > > >
> > > > For answering an offerless invite, I guess it could mean: offer
> > > > whatever
> > > > media you are capable of offering that match one of these, but
> > don't
> > > > offer anything that doesn't match.
> > > >
> > > > Is there likely to be a need to get more specific? E.g. about what
> > > > codecs to use? (Hopefully not - that could get very messy.)
> > > >
> > > >      o  transducer=3Dspeaker|headset
> > > >
> > > > My deskphone has three possibilities: speaker|headset|handset.
> > > > And I'm far from certain this is exhaustive. (The clue wg is
> > defining a
> > > > much more complex environment.) This at least needs to be
> > extensible.
> > > > We
> > > > may need some sort of extended model of a UA that includes this
> > sort of
> > > > stuff.
> > > >
> > > >      o  target=3D<AOR>
> > > >      o  direction=3Drecvonly|sendonly|sendrecv
> > > >      o  abort
> > > >
> > > > None of this is intended as objection to the work so far. Its just
> > a
> > > > start at identifying what else there is to do. The whole notion of
> > > > naming features to invoke causes me the most concern.
> > > >
> > > > 	Thanks,
> > > > 	Paul
> > > >
> > > > On 6/10/2011 3:08 PM, Shekh-Yusef, Rifaat (Rifaat) wrote:
> > > > > Hi,
> > > > >
> > > > > We have just submitted a new version of the SIP INVOKE method
> > draft
> > > > that
> > > > > does not include implicit subscription.
> > > > >
> > > > > https://datatracker.ietf.org/doc/draft-yusef-splices-invoke/
> > > > >
> > > > > We would appreciate it if you review the document and provide us
> > with
> > > > > your feedback.
> > > > >
> > > > > Thanks,
> > > > >
> > > > > Rifaat
> > > > >
> > > > >
> > > > >
> > > > > _______________________________________________
> > > > > splices mailing list
> > > > > splices@ietf.org
> > > > > https://www.ietf.org/mailman/listinfo/splices
> > > > _______________________________________________
> > > > splices mailing list
> > > > splices@ietf.org
> > > > https://www.ietf.org/mailman/listinfo/splices
> > > _______________________________________________
> > > splices mailing list
> > > splices@ietf.org
> > > https://www.ietf.org/mailman/listinfo/splices

From alan.b.johnston@gmail.com  Tue Jun 14 10:45:42 2011
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Date: Tue, 14 Jun 2011 12:45:41 -0500
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From: Alan Johnston <alan.b.johnston@gmail.com>
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Andy,

I agree with Dale that the set of actions should be small, otherwise,
we are doing something else.  Also, the set of parameters needs to be
small.  In fact, I think most actions won't have any parameters at
all.  This is why we are using URNs, just names, for the actions.

To do otherwise is to introduce complex actions or scripts, and I
don't want this mechanism to be used for that.  There are other
mechanisms (e.g. CPL) that allows a script to be uploaded or
downloaded to control a UA or proxy operation, but that is very
different from what we are talking about here.

I agree that we could discuss BNF vs XML, but if we agree that we
really are dealing with URNs, then a header makes more sense than a
body.

- Alan -

On Tue, Jun 14, 2011 at 12:32 PM, Worley, Dale R (Dale)
<dworley@avaya.com> wrote:
>> From: Hutton, Andrew [andrew.hutton@siemens-enterprise.com]
>>
>> The list of potential actions and associated parameters is huge and
>> for this kind of problem I believe that an XML body would be a much
>> better solution. I guess I much prefer XML to BNF.
>
> I'd feel much more comfortable if we settled on either:
>
> 1) a small number of actions
>
> 2) to use an appropriate subset of an existing CTI standard
>
> In the latter case, could someone provide a pointer to the standard?
>
> Essentially, INVOKE is to provide an interface to a command set, and
> we need to have some idea what the outer limits of the command set
> will be. =A0Only then can we choose a good way to represent the commands
> within a SIP request.
>
> Dale
> _______________________________________________
> splices mailing list
> splices@ietf.org
> https://www.ietf.org/mailman/listinfo/splices
>

From andrew.hutton@siemens-enterprise.com  Tue Jun 14 11:13:17 2011
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: "Worley, Dale R (Dale)" <dworley@avaya.com>, "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>, Paul Kyzivat <pkyzivat@cisco.com>, "splices@ietf.org" <splices@ietf.org>
Date: Tue, 14 Jun 2011 20:13:12 +0200
Thread-Topic: [splices] SIP INVOKE method v1
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Hi Dale,

I think the problem is that we don't have any specified requirements so we =
don't know what the boundaries are.=20

What has been specified so far already looks like it is trying to do CTI ty=
pe things and for that we know the list of possible services is endless. Th=
e most relevant standards would probably be ECMA-269 and ECMA-323 (http://w=
ww.ecma-international.org/publications/standards/Ecma-323.htm) if you want =
to see the XML for this.

I don't see how it is really possibly to limit this to a small set of actio=
ns as once the mechanism is implemented there will always be requirements t=
o add more actions.

Regards
Andy


> -----Original Message-----
> From: Worley, Dale R (Dale) [mailto:dworley@avaya.com]
> Sent: 14 June 2011 18:33
> To: Hutton, Andrew; Shekh-Yusef, Rifaat (Rifaat); Paul Kyzivat;
> splices@ietf.org
> Subject: RE: [splices] SIP INVOKE method v1
>=20
> > From: Hutton, Andrew [andrew.hutton@siemens-enterprise.com]
> >
> > The list of potential actions and associated parameters is huge and
> > for this kind of problem I believe that an XML body would be a much
> > better solution. I guess I much prefer XML to BNF.
>=20
> I'd feel much more comfortable if we settled on either:
>=20
> 1) a small number of actions
>=20
> 2) to use an appropriate subset of an existing CTI standard
>=20
> In the latter case, could someone provide a pointer to the standard?
>=20
> Essentially, INVOKE is to provide an interface to a command set, and
> we need to have some idea what the outer limits of the command set
> will be.  Only then can we choose a good way to represent the commands
> within a SIP request.
>=20
> Dale

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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: Alan Johnston <alan.b.johnston@gmail.com>, "Worley, Dale R (Dale)" <dworley@avaya.com>
Date: Tue, 14 Jun 2011 20:25:41 +0200
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Hi Alan,

Might be nice to keep it to a small set but I an not convinced it is possib=
le and most actions will need parameters when you consider different media =
types, audio devices etc. and even simple actions like call terminate will =
almost certainly need a reason code.

Regards
Andy


> -----Original Message-----
> From: Alan Johnston [mailto:alan.b.johnston@gmail.com]
> Sent: 14 June 2011 18:46
> To: Worley, Dale R (Dale)
> Cc: Hutton, Andrew; Shekh-Yusef, Rifaat (Rifaat); Paul Kyzivat;
> splices@ietf.org
> Subject: Re: [splices] SIP INVOKE method v1
>=20
> Andy,
>=20
> I agree with Dale that the set of actions should be small, otherwise,
> we are doing something else.  Also, the set of parameters needs to be
> small.  In fact, I think most actions won't have any parameters at
> all.  This is why we are using URNs, just names, for the actions.
>=20
> To do otherwise is to introduce complex actions or scripts, and I
> don't want this mechanism to be used for that.  There are other
> mechanisms (e.g. CPL) that allows a script to be uploaded or
> downloaded to control a UA or proxy operation, but that is very
> different from what we are talking about here.
>=20
> I agree that we could discuss BNF vs XML, but if we agree that we
> really are dealing with URNs, then a header makes more sense than a
> body.
>=20
> - Alan -
>=20
> On Tue, Jun 14, 2011 at 12:32 PM, Worley, Dale R (Dale)
> <dworley@avaya.com> wrote:
> >> From: Hutton, Andrew [andrew.hutton@siemens-enterprise.com]
> >>
> >> The list of potential actions and associated parameters is huge and
> >> for this kind of problem I believe that an XML body would be a much
> >> better solution. I guess I much prefer XML to BNF.
> >
> > I'd feel much more comfortable if we settled on either:
> >
> > 1) a small number of actions
> >
> > 2) to use an appropriate subset of an existing CTI standard
> >
> > In the latter case, could someone provide a pointer to the standard?
> >
> > Essentially, INVOKE is to provide an interface to a command set, and
> > we need to have some idea what the outer limits of the command set
> > will be. =A0Only then can we choose a good way to represent the
> commands
> > within a SIP request.
> >
> > Dale
> > _______________________________________________
> > splices mailing list
> > splices@ietf.org
> > https://www.ietf.org/mailman/listinfo/splices
> >

From dworley@avaya.com  Tue Jun 14 11:39:15 2011
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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>, "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>, Paul Kyzivat <pkyzivat@cisco.com>, "splices@ietf.org" <splices@ietf.org>
Date: Tue, 14 Jun 2011 14:35:48 -0400
Thread-Topic: [splices] SIP INVOKE method v1
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Subject: Re: [splices] SIP INVOKE method v1
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________________________________________
From: Hutton, Andrew [andrew.hutton@siemens-enterprise.com]

I think the problem is that we don't have any specified requirements so we =
don't know what the boundaries are.

What has been specified so far already looks like it is trying to do CTI ty=
pe things and for that we know the list of possible services is endless. Th=
e most relevant standards would probably be ECMA-269 and ECMA-323 (http://w=
ww.ecma-international.org/publications/standards/Ecma-323.htm) if you want =
to see the XML for this.

I don't see how it is really possibly to limit this to a small set of actio=
ns as once the mechanism is implemented there will always be requirements t=
o add more actions.
________________________________________

Thanks for those references!

Yes, there is a real scope-creep problem.  Looking at ECMA-323, I see 45 "C=
all control services and events", and that is not the majority of the defin=
ed operations.

One possibility would be to define INVOKE as a carrier for ECMA-323 operati=
ons.  But even then, a lot of care needs to be taken to see how to embed th=
e dataflow model of ECMA-323 into the SIP request/response structure.  If w=
e define our own CTI set, we have to create our own dataflow model.

Dale

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Subject: Re: [splices] SIP INVOKE method v1
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I'm with Dale that we need to scope the command set,
or equivalently, scope the environment that is being manipulated and the 
degrees of freedom that it has.

If it isn't "small" then it at least needs to be characterized. And the 
larger it is the bigger a deal the work is. So its important to figure 
it out. The mechanism to convey the commands is secondary to that.

For instance (as I have mentioned before in passing), if the scope 
includes "call control" in the sip sense, including o/a, then it becomes 
a very big deal unless we can reuse sip for it. We really don't want to 
define something new and different from sip that is functionally 
equivalent to sip.

My expectation is that we are talking about something at a higher level 
of abstraction than SIP. But being at a higher level doesn't mean 
"vague". And while it almost certainly needs to be extensible for things 
we don't consider at first, IMO that doesn't mean we should define a 
mechanism with no intrinsic functionality, just an extension mechanism. 
That is a guarantee we will get it wrong.

We can look to CSTA or one or more definitions of CTI. I did investigate 
CSTA once, and didn't love it. But if it comes close to the needs here 
we might consider adopting it. I don't know much about CTI, but my 
impression is that it is a buzz word with many only loosely related 
instantiations, and is predicated on different architectural models than 
will work for us. (But I'll be pleased to be educated by somebody who 
really knows.)

	Thanks,
	Paul

On 6/14/2011 2:25 PM, Hutton, Andrew wrote:
> Hi Alan,
>
> Might be nice to keep it to a small set but I an not convinced it is possible and most actions will need parameters when you consider different media types, audio devices etc. and even simple actions like call terminate will almost certainly need a reason code.
>
> Regards
> Andy
>
>
>> -----Original Message-----
>> From: Alan Johnston [mailto:alan.b.johnston@gmail.com]
>> Sent: 14 June 2011 18:46
>> To: Worley, Dale R (Dale)
>> Cc: Hutton, Andrew; Shekh-Yusef, Rifaat (Rifaat); Paul Kyzivat;
>> splices@ietf.org
>> Subject: Re: [splices] SIP INVOKE method v1
>>
>> Andy,
>>
>> I agree with Dale that the set of actions should be small, otherwise,
>> we are doing something else.  Also, the set of parameters needs to be
>> small.  In fact, I think most actions won't have any parameters at
>> all.  This is why we are using URNs, just names, for the actions.
>>
>> To do otherwise is to introduce complex actions or scripts, and I
>> don't want this mechanism to be used for that.  There are other
>> mechanisms (e.g. CPL) that allows a script to be uploaded or
>> downloaded to control a UA or proxy operation, but that is very
>> different from what we are talking about here.
>>
>> I agree that we could discuss BNF vs XML, but if we agree that we
>> really are dealing with URNs, then a header makes more sense than a
>> body.
>>
>> - Alan -
>>
>> On Tue, Jun 14, 2011 at 12:32 PM, Worley, Dale R (Dale)
>> <dworley@avaya.com>  wrote:
>>>> From: Hutton, Andrew [andrew.hutton@siemens-enterprise.com]
>>>>
>>>> The list of potential actions and associated parameters is huge and
>>>> for this kind of problem I believe that an XML body would be a much
>>>> better solution. I guess I much prefer XML to BNF.
>>>
>>> I'd feel much more comfortable if we settled on either:
>>>
>>> 1) a small number of actions
>>>
>>> 2) to use an appropriate subset of an existing CTI standard
>>>
>>> In the latter case, could someone provide a pointer to the standard?
>>>
>>> Essentially, INVOKE is to provide an interface to a command set, and
>>> we need to have some idea what the outer limits of the command set
>>> will be.  Only then can we choose a good way to represent the
>> commands
>>> within a SIP request.
>>>
>>> Dale
>>> _______________________________________________
>>> splices mailing list
>>> splices@ietf.org
>>> https://www.ietf.org/mailman/listinfo/splices
>>>
>

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From: Peter Musgrave <musgravepj@gmail.com>
Date: Tue, 14 Jun 2011 18:31:07 -0400
To: Paul Kyzivat <pkyzivat@cisco.com>
Cc: "splices@ietf.org" <splices@ietf.org>
Subject: Re: [splices] SIP INVOKE method v1
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I agree too.

I also thought CSTA was a bit too "kitchen sink" when I looked at it years a=
go. But perhaps we can use it to guide a very modest subset? It has at least=
 looked at a very similar problem space so there should be lessens learned t=
here...


Peter

Sent from my iPad

On 2011-06-14, at 3:56 PM, Paul Kyzivat <pkyzivat@cisco.com> wrote:

> I'm with Dale that we need to scope the command set,
> or equivalently, scope the environment that is being manipulated and the d=
egrees of freedom that it has.
>=20
> If it isn't "small" then it at least needs to be characterized. And the la=
rger it is the bigger a deal the work is. So its important to figure it out.=
 The mechanism to convey the commands is secondary to that.
>=20
> For instance (as I have mentioned before in passing), if the scope include=
s "call control" in the sip sense, including o/a, then it becomes a very big=
 deal unless we can reuse sip for it. We really don't want to define somethi=
ng new and different from sip that is functionally equivalent to sip.
>=20
> My expectation is that we are talking about something at a higher level of=
 abstraction than SIP. But being at a higher level doesn't mean "vague". And=
 while it almost certainly needs to be extensible for things we don't consid=
er at first, IMO that doesn't mean we should define a mechanism with no intr=
insic functionality, just an extension mechanism. That is a guarantee we wil=
l get it wrong.
>=20
> We can look to CSTA or one or more definitions of CTI. I did investigate C=
STA once, and didn't love it. But if it comes close to the needs here we mig=
ht consider adopting it. I don't know much about CTI, but my impression is t=
hat it is a buzz word with many only loosely related instantiations, and is p=
redicated on different architectural models than will work for us. (But I'll=
 be pleased to be educated by somebody who really knows.)
>=20
>    Thanks,
>    Paul
>=20
> On 6/14/2011 2:25 PM, Hutton, Andrew wrote:
>> Hi Alan,
>>=20
>> Might be nice to keep it to a small set but I an not convinced it is poss=
ible and most actions will need parameters when you consider different media=
 types, audio devices etc. and even simple actions like call terminate will a=
lmost certainly need a reason code.
>>=20
>> Regards
>> Andy
>>=20
>>=20
>>> -----Original Message-----
>>> From: Alan Johnston [mailto:alan.b.johnston@gmail.com]
>>> Sent: 14 June 2011 18:46
>>> To: Worley, Dale R (Dale)
>>> Cc: Hutton, Andrew; Shekh-Yusef, Rifaat (Rifaat); Paul Kyzivat;
>>> splices@ietf.org
>>> Subject: Re: [splices] SIP INVOKE method v1
>>>=20
>>> Andy,
>>>=20
>>> I agree with Dale that the set of actions should be small, otherwise,
>>> we are doing something else.  Also, the set of parameters needs to be
>>> small.  In fact, I think most actions won't have any parameters at
>>> all.  This is why we are using URNs, just names, for the actions.
>>>=20
>>> To do otherwise is to introduce complex actions or scripts, and I
>>> don't want this mechanism to be used for that.  There are other
>>> mechanisms (e.g. CPL) that allows a script to be uploaded or
>>> downloaded to control a UA or proxy operation, but that is very
>>> different from what we are talking about here.
>>>=20
>>> I agree that we could discuss BNF vs XML, but if we agree that we
>>> really are dealing with URNs, then a header makes more sense than a
>>> body.
>>>=20
>>> - Alan -
>>>=20
>>> On Tue, Jun 14, 2011 at 12:32 PM, Worley, Dale R (Dale)
>>> <dworley@avaya.com>  wrote:
>>>>> From: Hutton, Andrew [andrew.hutton@siemens-enterprise.com]
>>>>>=20
>>>>> The list of potential actions and associated parameters is huge and
>>>>> for this kind of problem I believe that an XML body would be a much
>>>>> better solution. I guess I much prefer XML to BNF.
>>>>=20
>>>> I'd feel much more comfortable if we settled on either:
>>>>=20
>>>> 1) a small number of actions
>>>>=20
>>>> 2) to use an appropriate subset of an existing CTI standard
>>>>=20
>>>> In the latter case, could someone provide a pointer to the standard?
>>>>=20
>>>> Essentially, INVOKE is to provide an interface to a command set, and
>>>> we need to have some idea what the outer limits of the command set
>>>> will be.  Only then can we choose a good way to represent the
>>> commands
>>>> within a SIP request.
>>>>=20
>>>> Dale
>>>> _______________________________________________
>>>> splices mailing list
>>>> splices@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/splices
>>>>=20
>>=20
> _______________________________________________
> splices mailing list
> splices@ietf.org
> https://www.ietf.org/mailman/listinfo/splices

From andrew.hutton@siemens-enterprise.com  Wed Jun 15 02:01:45 2011
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>, Paul Kyzivat <pkyzivat@cisco.com>, "splices@ietf.org" <splices@ietf.org>
Date: Wed, 15 Jun 2011 11:01:38 +0200
Thread-Topic: [splices] SIP INVOKE method v1
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Subject: Re: [splices] SIP INVOKE method v1
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Hi Rifaat,

I don't understand the argument as the header/body issue makes no differenc=
e to what gets standardized and architecturally it is much cleaner to tunne=
l the non SIP related application stuff in a body that to use a SIP header =
for the same purpose.

Having said that I still don't think that either mechanism would fall withi=
n the SPLICES charter.

Regards
Andy



> -----Original Message-----
> From: Shekh-Yusef, Rifaat (Rifaat) [mailto:rifatyu@avaya.com]
> Sent: 14 June 2011 18:42
> To: Hutton, Andrew; Paul Kyzivat; splices@ietf.org
> Subject: RE: [splices] SIP INVOKE method v1
>=20
> Andy,
>=20
> > The list of potential actions and associated parameters is huge and
> for this
> > kind of problem I believe that an XML body would be a much better
> solution. I
> > guess I much prefer XML to BNF.
> >
> This actually calls for a header rather than an XML body, because the
> idea is to control what get standardized and what not and clearly
> define the actions that get standardized. With an XML body you actually
> create a tunnel over SIP and you lose control over what get defined.
>=20
> Regards,
>  Rifaat
>=20
>=20
> > -----Original Message-----
> > From: Hutton, Andrew [mailto:andrew.hutton@siemens-enterprise.com]
> > Sent: Tuesday, June 14, 2011 1:18 PM
> > To: Shekh-Yusef, Rifaat (Rifaat); Paul Kyzivat; splices@ietf.org
> > Subject: RE: [splices] SIP INVOKE method v1
> >
> > Hi Rifaat,
> >
> > I am not saying that the current proposal is not extensible but I
> believe that
> > an XML body would be a much better solution to the extensibility
> problem that
> > using a SIP header.
> >
> > The list of potential actions and associated parameters is huge and
> for this
> > kind of problem I believe that an XML body would be a much better
> solution. I
> > guess I much prefer XML to BNF.
> >
> > There is also an architectural issue as the actions don't relate
> directly to
> > the management of the SIP session and therefore most likely need to
> passed to
> > the application layer in the SIP UA to handle. It is much cleaner
> > architecturally if the body can simply be removed and passed to the
> > responsible component in the UA rather than the SIP component having
> to
> > extract the information from SIP headers.
> >
> > Regards
> > Andy
> >
> >
> >
> >
> > > -----Original Message-----
> > > From: Shekh-Yusef, Rifaat (Rifaat) [mailto:rifatyu@avaya.com]
> > > Sent: 14 June 2011 17:32
> > > To: Hutton, Andrew; Paul Kyzivat; splices@ietf.org
> > > Subject: RE: [splices] SIP INVOKE method v1
> > >
> > > Andy,
> > >
> > > > The issue is I believe extensibility which any mechanism for
> > > > invoking services will need and this is the reason why an XML
> body
> > > would be
> > > > better than using SIP headers.
> > >
> > > What makes you think that the current proposal is not extensible?
> > >
> > > Regards,
> > >  Rifaat
> > >
> > >
> > > > -----Original Message-----
> > > > From: splices-bounces@ietf.org [mailto:splices-bounces@ietf.org]
> On
> > > Behalf Of
> > > > Hutton, Andrew
> > > > Sent: Monday, June 13, 2011 10:04 AM
> > > > To: Paul Kyzivat; splices@ietf.org
> > > > Subject: Re: [splices] SIP INVOKE method v1
> > > >
> > > > Hi All,
> > > >
> > > > Whilst I still question whether the mechanism described in this
> draft
> > > actually
> > > > falls within the SPLICES WG charter as it defines new mechanism I
> > > have a few
> > > > comments.
> > > >
> > > > I tend to agree with Paul's comments below but would go further
> to
> > > say that
> > > > even some of the supposedly simply actions such as "decline" &
> > > "ignore" are
> > > > also features which will be difficult to agree the semantics for.
> All
> > > of the
> > > > actions will need to be specified rigorously a quick look at
> ECMA-269
> > > which
> > > > specifies these kind of services for CSTA will give you an idea
> of
> > > what would
> > > > need to be done. I don't believe that even in the CSTA world
> which
> > > defines a
> > > > very large number of services the services "ignore" or "decline"
> were
> > > > specified.
> > > >
> > > > With regard to the discussion on headers vs body the draft
> states.
> > > >
> > > > "There has been some discussion on the list on the question of
> header
> > > field vs
> > > > message body.  This is not the real issue, as either could
> > > conceivably be
> > > > used.  Instead, the main issue is what is being invoked:  is it a
> > > named
> > > > feature/action/event, or is it a script".
> > > >
> > > > I must have missed the discussion about scripting but I don't
> think
> > > this is
> > > > the issue. The issue is I believe extensibility which any
> mechanism
> > > for
> > > > invoking services will need and this is the reason why an XML
> body
> > > would be
> > > > better than using SIP headers.
> > > >
> > > > Regards
> > > > Andy
> > > >
> > > >
> > > > > -----Original Message-----
> > > > > From: splices-bounces@ietf.org [mailto:splices-
> bounces@ietf.org] On
> > > > > Behalf Of Paul Kyzivat
> > > > > Sent: 10 June 2011 20:46
> > > > > To: splices@ietf.org
> > > > > Subject: Re: [splices] SIP INVOKE method v1
> > > > >
> > > > > This is getting better with age!
> > > > >
> > > > > Splitting off the subscription is good.
> > > > >
> > > > > An area that still needs work is in the semantics of the urns.
> > > > > Some of them are more clear than others. Some are "limited" in
> ways
> > > > > that
> > > > > might not matter to some but that could prove a problem in the
> end.
> > > > > Specifically:
> > > > >
> > > > >      o  Answer call       - urn:invoke:call:answer
> > > > >      o  Terminate call    - urn:invoke:call:terminate
> > > > >      o  Decline call      - urn:invoke:call:decline
> > > > >      o  Ignore call       - urn:invoke:call:ignore
> > > > >
> > > > > The above need some elaboration, but I think its just work -
> > > probably
> > > > > anybody who does it will end up with an equivalent result.
> > > > >
> > > > >      o  Send call to VM   - urn:invoke:call:sendvm
> > > > >
> > > > > VM is a "feature", and we haven't standardized features much,
> > > except
> > > > > for
> > > > > bliss. How does one know if the recipient understands VM and
> what
> > > to do
> > > > > to send a call there? What if there are multiple VM options
> (e.g.
> > > what
> > > > > to say, etc.) And what if the call isn't a voice call. (Maybe
> VM
> > > now
> > > > > means voice/video mail. But the "call" could be IM.)
> > > > >
> > > > >      o  Hold call         - urn:invoke:call:hold
> > > > >      o  Unhold call       - urn:invoke:call:unhold
> > > > >      o  Mute call         - urn:invoke:call:mute
> > > > >      o  Unmute call       - urn:invoke:call:unmute
> > > > >
> > > > > Similarly these are all "features" that we all have some
> > > expectations
> > > > > about, but that are not rigorously defined. (But perhaps
> > > Mute/Unmute
> > > > > could be formalized by using the 'transducer' option.
> > > > > (Transducer=3Dnone?)
> > > > >
> > > > > My concern with all these "features" is that we are being put
> into
> > > a
> > > > > box
> > > > > where we must either:
> > > > > - define them with sufficient rigor
> > > > > - fall back on "do what I mean", which has proved to be a bad
> idea,
> > > > >    because not everybody has the same notion of what they mean.
> > > > >
> > > > >      o  Conference        - urn:invoke:conference:add
> > > > >                           - urn:invoke:conference:remove
> > > > >
> > > > > This one may be clearer due to the existence of xcon specs.
> > > > >
> > > > > Regarding action parameters:
> > > > >
> > > > >      o  media=3Daudio|video|audvid
> > > > >
> > > > > I'm troubled by audvid. I would rather see "audio,video".
> > > Specifically,
> > > > > what is the namespace here, and what does specifying it mean? I
> > > think
> > > > > the namespace could be: one or more values allowable as the
> <media>
> > > > > parameter in the m-line in SDP.
> > > > >
> > > > > Defining exactly what it means when present is a little harder.
> > > When
> > > > > answering a call that has an offer, I guess it could mean:
> refuse
> > > any
> > > > > m-lines with media type that doesn't match one of these, and
> accept
> > > as
> > > > > many of the remaining as you can.
> > > > >
> > > > > For answering an offerless invite, I guess it could mean: offer
> > > > > whatever
> > > > > media you are capable of offering that match one of these, but
> > > don't
> > > > > offer anything that doesn't match.
> > > > >
> > > > > Is there likely to be a need to get more specific? E.g. about
> what
> > > > > codecs to use? (Hopefully not - that could get very messy.)
> > > > >
> > > > >      o  transducer=3Dspeaker|headset
> > > > >
> > > > > My deskphone has three possibilities: speaker|headset|handset.
> > > > > And I'm far from certain this is exhaustive. (The clue wg is
> > > defining a
> > > > > much more complex environment.) This at least needs to be
> > > extensible.
> > > > > We
> > > > > may need some sort of extended model of a UA that includes this
> > > sort of
> > > > > stuff.
> > > > >
> > > > >      o  target=3D<AOR>
> > > > >      o  direction=3Drecvonly|sendonly|sendrecv
> > > > >      o  abort
> > > > >
> > > > > None of this is intended as objection to the work so far. Its
> just
> > > a
> > > > > start at identifying what else there is to do. The whole notion
> of
> > > > > naming features to invoke causes me the most concern.
> > > > >
> > > > > 	Thanks,
> > > > > 	Paul
> > > > >
> > > > > On 6/10/2011 3:08 PM, Shekh-Yusef, Rifaat (Rifaat) wrote:
> > > > > > Hi,
> > > > > >
> > > > > > We have just submitted a new version of the SIP INVOKE method
> > > draft
> > > > > that
> > > > > > does not include implicit subscription.
> > > > > >
> > > > > > https://datatracker.ietf.org/doc/draft-yusef-splices-invoke/
> > > > > >
> > > > > > We would appreciate it if you review the document and provide
> us
> > > with
> > > > > > your feedback.
> > > > > >
> > > > > > Thanks,
> > > > > >
> > > > > > Rifaat
> > > > > >
> > > > > >
> > > > > >
> > > > > > _______________________________________________
> > > > > > splices mailing list
> > > > > > splices@ietf.org
> > > > > > https://www.ietf.org/mailman/listinfo/splices
> > > > > _______________________________________________
> > > > > splices mailing list
> > > > > splices@ietf.org
> > > > > https://www.ietf.org/mailman/listinfo/splices
> > > > _______________________________________________
> > > > splices mailing list
> > > > splices@ietf.org
> > > > https://www.ietf.org/mailman/listinfo/splices

From R.Jesske@telekom.de  Thu Jun 16 01:52:16 2011
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Date: Thu, 16 Jun 2011 10:50:50 +0200
Thread-Topic: [splices] SIP INVOKE method v1
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Subject: Re: [splices] SIP INVOKE method v1
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Hi,
now I'm puzzled about the scope?

Is the scope of INVOKE to realise the Computer Supported Telecommunications=
 Applications (CSTA)?

Best Regards
Roland




> -----Urspr=FCngliche Nachricht-----
> Von: splices-bounces@ietf.org
> [mailto:splices-bounces@ietf.org] Im Auftrag von Worley, Dale R (Dale)
> Gesendet: Dienstag, 14. Juni 2011 20:36
> An: Hutton, Andrew; Shekh-Yusef, Rifaat (Rifaat); Paul
> Kyzivat; splices@ietf.org
> Betreff: Re: [splices] SIP INVOKE method v1
>
> ________________________________________
> From: Hutton, Andrew [andrew.hutton@siemens-enterprise.com]
>
> I think the problem is that we don't have any specified
> requirements so we don't know what the boundaries are.
>
> What has been specified so far already looks like it is
> trying to do CTI type things and for that we know the list of
> possible services is endless. The most relevant standards
> would probably be ECMA-269 and ECMA-323
> (http://www.ecma-international.org/publications/standards/Ecma
> -323.htm) if you want to see the XML for this.
>
> I don't see how it is really possibly to limit this to a
> small set of actions as once the mechanism is implemented
> there will always be requirements to add more actions.
> ________________________________________
>
> Thanks for those references!
>
> Yes, there is a real scope-creep problem.  Looking at
> ECMA-323, I see 45 "Call control services and events", and
> that is not the majority of the defined operations.
>
> One possibility would be to define INVOKE as a carrier for
> ECMA-323 operations.  But even then, a lot of care needs to
> be taken to see how to embed the dataflow model of ECMA-323
> into the SIP request/response structure.  If we define our
> own CTI set, we have to create our own dataflow model.
>
> Dale
> _______________________________________________
> splices mailing list
> splices@ietf.org
> https://www.ietf.org/mailman/listinfo/splices
>

From dworley@avaya.com  Fri Jun 17 13:34:51 2011
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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>, "splices@ietf.org" <splices@ietf.org>
Date: Fri, 17 Jun 2011 16:34:49 -0400
Thread-Topic: [splices] Initiate an A/V Call Using Two Separate Devices proposal
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Subject: Re: [splices] Initiate an A/V Call Using Two Separate Devices proposal
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I like the general idea of this, but I'm not fond of the details.

urn:invoke:call:initiate;media=3Dvideo;target=3Dbob@whatever
seems to mean, "Initiate a call to bob@whatever, adding me as a 3rd-party o=
rigin for video".

But I don't see the need for the "media=3Dvideo" part of the URN, as the co=
ntroller
will have the opportunity to edit my SDP when it receives it before passing=
 it on.

But the URN urn:invoke:call:initiate;target=3Dbob@whatever would seem to te=
ll
the UAS to initiate a call to bob@whatever, without including the UAC as a =
3pcc participant.

urn:invoke:call:initiate;media=3Dvideo
seems to mean, "Initiate a call to me, offering me your video resources."

But again, there seems to be no reason to specify which media are desired, =
as the UAC
can reject any offered streams that it doesn't like.

I think we want to avoid having to specify media types in the URNs whenever=
 possible, as they
will be multiplying over time.  How would one specify, "Offer me your audio=
 and text resources?"

Dale

From rifatyu@avaya.com  Sat Jun 18 06:23:56 2011
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From: "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>
To: "R.Jesske@telekom.de" <R.Jesske@telekom.de>, "Worley, Dale R (Dale)" <dworley@avaya.com>, "andrew.hutton@siemens-enterprise.com" <andrew.hutton@siemens-enterprise.com>, "pkyzivat@cisco.com" <pkyzivat@cisco.com>, "splices@ietf.org" <splices@ietf.org>
Date: Sat, 18 Jun 2011 09:23:52 -0400
Thread-Topic: [splices] SIP INVOKE method v1
Thread-Index: AcwnpwHyWdPZ0GtcQdet0Lc+UVQPIgCJvnyQADiYGNAAANdDQAABWEfrAAFc1lAAANTSAwBPvZmgADtjeYA=
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Hi Roland,

I browsed through the CSTA documents. This seems to go way beyond what we h=
ad in mind for the actions with the INVOKE method.
I will shortly send a separate email to start a discussion around the scope=
 of the actions with the INVOKE method.

Regards,
 Rifaat


> -----Original Message-----
> From: R.Jesske@telekom.de [mailto:R.Jesske@telekom.de]
> Sent: Thursday, June 16, 2011 4:51 AM
> To: Worley, Dale R (Dale); andrew.hutton@siemens-enterprise.com; Shekh-Yu=
sef,
> Rifaat (Rifaat); pkyzivat@cisco.com; splices@ietf.org
> Subject: AW: [splices] SIP INVOKE method v1
>=20
> Hi,
> now I'm puzzled about the scope?
>=20
> Is the scope of INVOKE to realise the Computer Supported Telecommunicatio=
ns
> Applications (CSTA)?
>=20
> Best Regards
> Roland
>=20
>=20
>=20
>=20
> > -----Urspr=FCngliche Nachricht-----
> > Von: splices-bounces@ietf.org
> > [mailto:splices-bounces@ietf.org] Im Auftrag von Worley, Dale R (Dale)
> > Gesendet: Dienstag, 14. Juni 2011 20:36
> > An: Hutton, Andrew; Shekh-Yusef, Rifaat (Rifaat); Paul
> > Kyzivat; splices@ietf.org
> > Betreff: Re: [splices] SIP INVOKE method v1
> >
> > ________________________________________
> > From: Hutton, Andrew [andrew.hutton@siemens-enterprise.com]
> >
> > I think the problem is that we don't have any specified
> > requirements so we don't know what the boundaries are.
> >
> > What has been specified so far already looks like it is
> > trying to do CTI type things and for that we know the list of
> > possible services is endless. The most relevant standards
> > would probably be ECMA-269 and ECMA-323
> > (http://www.ecma-international.org/publications/standards/Ecma
> > -323.htm) if you want to see the XML for this.
> >
> > I don't see how it is really possibly to limit this to a
> > small set of actions as once the mechanism is implemented
> > there will always be requirements to add more actions.
> > ________________________________________
> >
> > Thanks for those references!
> >
> > Yes, there is a real scope-creep problem.  Looking at
> > ECMA-323, I see 45 "Call control services and events", and
> > that is not the majority of the defined operations.
> >
> > One possibility would be to define INVOKE as a carrier for
> > ECMA-323 operations.  But even then, a lot of care needs to
> > be taken to see how to embed the dataflow model of ECMA-323
> > into the SIP request/response structure.  If we define our
> > own CTI set, we have to create our own dataflow model.
> >
> > Dale
> > _______________________________________________
> > splices mailing list
> > splices@ietf.org
> > https://www.ietf.org/mailman/listinfo/splices
> >=

From rifatyu@avaya.com  Sat Jun 18 06:37:24 2011
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From: "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>
To: "splices@ietf.org" <splices@ietf.org>
Date: Sat, 18 Jun 2011 09:37:22 -0400
Thread-Topic: [splices] INVOKE Actions Scope
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Hi,

I would like to start a discussion that I hope would allow us to converge o=
n an approriate scope for the first release of this draft.
The following is my first attempt at defining this scope:

1. An Action can manipulate a dialog or a multimedia session.
2. An Action can manipulate one or more dialogs or sessions at the same tim=
e.
3. An Action can manipulate the state of a physical "feature" of a device, =
e.g. transducer.
4. An Action can invoke a well-defined telephony "feature" on the UA, e.g.h=
old.
5. An Action cannot replace an existing SIP mechanism, e.g. offer/answer.
6. An Action cannot dictate the UI behavior of the target UA.

Any thoughts?

Thanks,
Rifaat=

From rifatyu@avaya.com  Sat Jun 18 06:44:10 2011
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From: "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>
To: "Worley, Dale R (Dale)" <dworley@avaya.com>, "splices@ietf.org" <splices@ietf.org>
Date: Sat, 18 Jun 2011 09:44:07 -0400
Thread-Topic: [splices] Initiate an A/V Call Using Two Separate Devices proposal
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Hi Dale,

Good point.
I am fine with your proposal of not specifying the media type in a URN.

Regards,
 Rifaat

________________________________________
From: Worley, Dale R (Dale)
Sent: Friday, June 17, 2011 4:34 PM
To: Shekh-Yusef, Rifaat (Rifaat); splices@ietf.org
Subject: RE: [splices] Initiate an A/V Call Using Two Separate Devices prop=
osal

I like the general idea of this, but I'm not fond of the details.

urn:invoke:call:initiate;media=3Dvideo;target=3Dbob@whatever
seems to mean, "Initiate a call to bob@whatever, adding me as a 3rd-party o=
rigin for video".

But I don't see the need for the "media=3Dvideo" part of the URN, as the co=
ntroller
will have the opportunity to edit my SDP when it receives it before passing=
 it on.

But the URN urn:invoke:call:initiate;target=3Dbob@whatever would seem to te=
ll
the UAS to initiate a call to bob@whatever, without including the UAC as a =
3pcc participant.

urn:invoke:call:initiate;media=3Dvideo
seems to mean, "Initiate a call to me, offering me your video resources."

But again, there seems to be no reason to specify which media are desired, =
as the UAC
can reject any offered streams that it doesn't like.

I think we want to avoid having to specify media types in the URNs whenever=
 possible, as they
will be multiplying over time.  How would one specify, "Offer me your audio=
 and text resources?"

Dale=

From pkyzivat@cisco.com  Sat Jun 18 10:40:45 2011
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Subject: Re: [splices] INVOKE Actions Scope
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On 6/18/2011 9:37 AM, Shekh-Yusef, Rifaat (Rifaat) wrote:
> Hi,
>
> I would like to start a discussion that I hope would allow us to converge on an approriate scope for the first release of this draft.
> The following is my first attempt at defining this scope:
>
> 1. An Action can manipulate a dialog or a multimedia session.
> 2. An Action can manipulate one or more dialogs or sessions at the same time.

"Manipulate" is a fuzzy word.

You can look in 3261 and *start* to identify the state of a dialog.
Then you can look at 5057 for more on dialog usages.

If we were to consider any manipulation of any dialog or dialog usage 
state, we would need to support complex arguments - SDP, maybe event 
payloads, etc.

IMO we need to scope down more. E.g. only invite-dialog-usages, and the 
kinds of manipulation can't involve getting into fine details. We need 
to find a way to abstract away the details. Either we create our own 
abstraction, or else we adopt some existing abstraction. Unfortunately, 
I fear the existing abstractions may not cover multimedia sufficiently 
for our needs.

> 3. An Action can manipulate the state of a physical "feature" of a device, e.g. transducer.

"Transducer" seems like it might be a useful abstraction here.
But again, we need to understand it a little better to be able to 
understand what sort of "manipulation" is possible.

ISTM that "transducers" are the sources and sinks of media streams.
The class of manipulations we care about is perhaps just the 
attachment/detachment of transducers to media streams. Does that seem right?

> 4. An Action can invoke a well-defined telephony "feature" on the UA, e.g.hold.

I'm not so certain how "well-defined" these are, especially in the sip 
context.

E.g., does "hold" apply only to audio? Or does it apply in parallel to 
all media streams? Or can you hold individual streams? Is it 
music/video/etc on hold, or simply stopping the media?

IMO even "well known features" need to be individually defined.
This starts to feel like what bliss was doing.

> 5. An Action cannot replace an existing SIP mechanism, e.g. offer/answer.
> 6. An Action cannot dictate the UI behavior of the target UA.
>
> Any thoughts?
>
> Thanks,
> Rifaat
> _______________________________________________
> splices mailing list
> splices@ietf.org
> https://www.ietf.org/mailman/listinfo/splices
>

From salvatore.loreto@ericsson.com  Sun Jun 19 06:12:48 2011
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Subject: Re: [splices] Initiate an A/V Call Using Two Separate Devices proposal
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while I agree with Dale that for the specific URN (i.e.: 
urn:invoke:call:initiate;)
used in the example does not make sense specifying the media type,
I am not so sure that not specifying the media type in a URN is the 
right thing to do in general

cheers
/Sal


On 6/18/11 4:44 PM, Shekh-Yusef, Rifaat (Rifaat) wrote:
> Hi Dale,
>
> Good point.
> I am fine with your proposal of not specifying the media type in a URN.
>
> Regards,
>   Rifaat
>
> ________________________________________
> From: Worley, Dale R (Dale)
> Sent: Friday, June 17, 2011 4:34 PM
> To: Shekh-Yusef, Rifaat (Rifaat); splices@ietf.org
> Subject: RE: [splices] Initiate an A/V Call Using Two Separate Devices proposal
>
> I like the general idea of this, but I'm not fond of the details.
>
> urn:invoke:call:initiate;media=video;target=bob@whatever
> seems to mean, "Initiate a call to bob@whatever, adding me as a 3rd-party origin for video".
>
> But I don't see the need for the "media=video" part of the URN, as the controller
> will have the opportunity to edit my SDP when it receives it before passing it on.
>
> But the URN urn:invoke:call:initiate;target=bob@whatever would seem to tell
> the UAS to initiate a call to bob@whatever, without including the UAC as a 3pcc participant.
>
> urn:invoke:call:initiate;media=video
> seems to mean, "Initiate a call to me, offering me your video resources."
>
> But again, there seems to be no reason to specify which media are desired, as the UAC
> can reject any offered streams that it doesn't like.
>
> I think we want to avoid having to specify media types in the URNs whenever possible, as they
> will be multiplying over time.  How would one specify, "Offer me your audio and text resources?"
>
> Dale
> _______________________________________________
> splices mailing list
> splices@ietf.org
> https://www.ietf.org/mailman/listinfo/splices



From gonzalo.camarillo@ericsson.com  Tue Jun 21 01:10:08 2011
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Subject: Re: [splices] Using Two Separate Devices to Start a Conversation proposal
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Hi Alan,

>> As Dale mentioned, this also depends on the other end doing the "right
>> thing" with this case. It is a leap of faith to assume it will realize it
>> should accept both streams and use one for input and the other for output.
>>
>> ISTM its at least as likely that it would accept the first stream, reject
>> the second, and then have only a one-way "conversation".
> 
> OK, I understand the problem right now.  The problem is that this is a
> special kind of "Join" operation.  It is kind of a join, but not
> quite.
> 
> The operation we are after is perhaps similar to 3.3.8 Far-Fork in RFC 5850:
> 
> I suspect instead of Join, we need a new primitive, lets call it
> Splice or Merge which requests the UAS to add the media in the INVITE
> into the dialog that is referenced.

Yes... this is exactly what we proposed 5 years ago in one of the drafts
that, eventually, led to the creation of the SPLICES WG:

http://tools.ietf.org/html/draft-loreto-sipping-dialog-correlation-01

The idea was that the media streams of a given session could be
established using more than one SIP dialog.

Note that most SPLICES-related discussions up to a couple of months ago
were based on the fact that such a mechanism (i.e., using different
dialogs to establish a single session) would require support from the
remote end point. That is also the point Paul made in his comment above.
I personally would be OK with a mechanism along these lines, though.

Cheers,

Gonzalo


From gonzalo.camarillo@ericsson.com  Tue Jun 21 01:31:45 2011
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Subject: [splices] Fwd: [RAI] Device control using SIP
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Hi,

given the amount of discussions on the INVOKE method in this list, I
sent the email below to the RAI list to get the opinion of the whole
community. I would like to have this discussion sooner rather than later
so that we have all the information we need related to this piece of work.

Please, follow those discussions (and send your opinion) on the RAI list.

Cheers,

Gonzalo

-------- Original Message --------
Subject: [RAI] Device control using SIP
Date: Tue, 21 Jun 2011 10:27:57 +0200
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
To: rai@ietf.org <rai@ietf.org>

Hi,

the SPLICES WG is chartered to work on a framework for disaggregated
media in SIP.

https://datatracker.ietf.org/wg/splices/charter/

As part of the SPLICES framework, the WG has been discussing a SIP-based
mechanism to control devices.

https://datatracker.ietf.org/doc/draft-yusef-splices-invoke/

As you can see, this is an individual submission and the SPLICES WG
still has not decided whether or not this will be needed. However, this
draft has attracted a significant amount of attention on the SPLICES
mailing list.

The reason for this "heads up" is that this issue has been discussed at
length by the RAI community in the past. You may remember the "explode
my phone" discussions where a user agent sent such a command to another
one :o)

I would like to see an explicit discussion on whether or not this type
of application (i.e., device control) should fall within the scope of SIP.

Thanks,

Gonzalo

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


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Cc: "splices@ietf.org" <splices@ietf.org>
Subject: Re: [splices] Using Two Separate Devices to Start a Conversation proposal
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IMO there is nothing wrong with using Join to bring together one dialog 
with audio and another with video. I don't see why we would create 
another mechanism when that one is there to be used. Certainly it is a 
degenerate case of Join, but there is nothing wrong with degenerate cases.

Of course there might be UAs that are capable of supporting the case 
where the media in the dialogs is disjoint, but not the case where there 
is common media in the dialogs that must be mixed. But surely we can 
find an error code to cover that case.

The key issue here is what UA we expect to support this capability. IMO 
a user that wants to "splice" some of its devices together should take 
the responsibility of finding a UA that can handle the coordination, 
rather than depending upon the far end to do it.

	Thanks,
	Paul

On 6/21/2011 4:09 AM, Gonzalo Camarillo wrote:
> Hi Alan,
>
>>> As Dale mentioned, this also depends on the other end doing the "right
>>> thing" with this case. It is a leap of faith to assume it will realize it
>>> should accept both streams and use one for input and the other for output.
>>>
>>> ISTM its at least as likely that it would accept the first stream, reject
>>> the second, and then have only a one-way "conversation".
>>
>> OK, I understand the problem right now.  The problem is that this is a
>> special kind of "Join" operation.  It is kind of a join, but not
>> quite.
>>
>> The operation we are after is perhaps similar to 3.3.8 Far-Fork in RFC 5850:
>>
>> I suspect instead of Join, we need a new primitive, lets call it
>> Splice or Merge which requests the UAS to add the media in the INVITE
>> into the dialog that is referenced.
>
> Yes... this is exactly what we proposed 5 years ago in one of the drafts
> that, eventually, led to the creation of the SPLICES WG:
>
> http://tools.ietf.org/html/draft-loreto-sipping-dialog-correlation-01
>
> The idea was that the media streams of a given session could be
> established using more than one SIP dialog.
>
> Note that most SPLICES-related discussions up to a couple of months ago
> were based on the fact that such a mechanism (i.e., using different
> dialogs to establish a single session) would require support from the
> remote end point. That is also the point Paul made in his comment above.
> I personally would be OK with a mechanism along these lines, though.
>
> Cheers,
>
> Gonzalo
>
>

From fluffy@cisco.com  Wed Jun 22 10:11:23 2011
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Subject: Re: [splices] INVOKE Actions Scope
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There are a few features that most PBX can do today there is no =
standardized way to use SIP to signal the changes to sip dialog state =
needed to implement these features.  I think we should pick an initial =
small set of features that are often used and supported and do theses - =
that was what I hoped this draft did. It proposes a few things, probably =
some need to be specified in more detail, perhaps there are a few more =
that people think should be added, perhaps a few proposed should be =
removed.=20

So my proposal for scope is just the scope we have in this draft which I =
would argue is a subset of the "An Action can manipulate a dialog or a =
multimedia session." I view all the action here as changing the SIP =
dialog state and what changes happen to the multimedia session being a =
side effect of that.=20

On Jun 18, 2011, at 6:37 , Shekh-Yusef, Rifaat (Rifaat) wrote:

> Hi,
>=20
> I would like to start a discussion that I hope would allow us to =
converge on an approriate scope for the first release of this draft.
> The following is my first attempt at defining this scope:
>=20
> 1. An Action can manipulate a dialog or a multimedia session.
> 2. An Action can manipulate one or more dialogs or sessions at the =
same time.
> 3. An Action can manipulate the state of a physical "feature" of a =
device, e.g. transducer.
> 4. An Action can invoke a well-defined telephony "feature" on the UA, =
e.g.hold.
> 5. An Action cannot replace an existing SIP mechanism, e.g. =
offer/answer.
> 6. An Action cannot dictate the UI behavior of the target UA.
>=20
> Any thoughts?
>=20
> Thanks,
> Rifaat
> _______________________________________________
> splices mailing list
> splices@ietf.org
> https://www.ietf.org/mailman/listinfo/splices


Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html



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Cullen,

On 6/22/2011 1:11 PM, Cullen Jennings wrote:
>
> There are a few features that most PBX can do today there is no standardized way to use SIP to signal the changes to sip dialog state needed to implement these features.  I think we should pick an initial small set of features that are often used and supported and do theses - that was what I hoped this draft did. It proposes a few things, probably some need to be specified in more detail, perhaps there are a few more that people think should be added, perhaps a few proposed should be removed.
>
> So my proposal for scope is just the scope we have in this draft which I would argue is a subset of the "An Action can manipulate a dialog or a multimedia session." I view all the action here as changing the SIP dialog state and what changes happen to the multimedia session being a side effect of that.

The places where I think this wants/needs to go which is *related* to 
dialogs but which fundamentally expands the scope are:

- control of "transducers". Its not rocket science, but its new, and
   its going to require an architectural model of what transducers are
   and how they relate to the existing stuff in sip

- invoking of "features", such as "send the call to voicemail".
   That is at a higher level of abstraction than dialogs. Either
   we have to wave our hands and say "do what I mean", or we need
   to actually define what we mean. I realize Bliss is done and
   gone, but this "smells" like the kind of thing bliss was doing.

	Thanks,
	Paul

> On Jun 18, 2011, at 6:37 , Shekh-Yusef, Rifaat (Rifaat) wrote:
>
>> Hi,
>>
>> I would like to start a discussion that I hope would allow us to converge on an approriate scope for the first release of this draft.
>> The following is my first attempt at defining this scope:
>>
>> 1. An Action can manipulate a dialog or a multimedia session.
>> 2. An Action can manipulate one or more dialogs or sessions at the same time.
>> 3. An Action can manipulate the state of a physical "feature" of a device, e.g. transducer.
>> 4. An Action can invoke a well-defined telephony "feature" on the UA, e.g.hold.
>> 5. An Action cannot replace an existing SIP mechanism, e.g. offer/answer.
>> 6. An Action cannot dictate the UI behavior of the target UA.
>>
>> Any thoughts?
>>
>> Thanks,
>> Rifaat
>> _______________________________________________
>> splices mailing list
>> splices@ietf.org
>> https://www.ietf.org/mailman/listinfo/splices
>
>
> Cullen Jennings
> For corporate legal information go to:
> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>
>
> _______________________________________________
> splices mailing list
> splices@ietf.org
> https://www.ietf.org/mailman/listinfo/splices
>

From spromano@unina.it  Fri Jun 24 15:11:00 2011
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Inizio messaggio inoltrato:

> Da: NomCom Chair <nomcom-chair@ietf.org>
> Data: 24 giugno 2011 20:35:13 GMT+02:00
> A: Working Group Chairs <wgchairs@ietf.org>
> Oggetto: Please help the Nomcom 
> 

> Hi WG chairs,
>  We have had a good response to the first call for volunteers but the 
> rate at which new volunteers are coming in is slowing down. The Nomcom 
> process is best served by a large pool of volunteers drawn from a wide 
> spectrum of IETF attendees. Where else would we find this wide spectrum
> if not in the WG mailing lists.
> 
> I would really appreciate it if you can forward the message onto your
> working group mailing lists. 
> 
> The latest volunteer status and the second call for volunteers can be
> found at
> https://datatracker.ietf.org/ann/nomcom/2964/
> 
> Thanks in advance for your help.
> 
> Suresh Krishnan
> Nomcom Chair 2011-2012
> Email: nomcom-chair@ietf.org, suresh.krishnan@ericsson.com
> 

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<html><body bgcolor=3D"#FFFFFF"><div><br><br><br>Inizio messaggio inoltrato:=
<br><br></div><blockquote type=3D"cite"><div><b>Da:</b> NomCom Chair &lt;<a h=
ref=3D"mailto:nomcom-chair@ietf.org">nomcom-chair@ietf.org</a>&gt;<br><b>Dat=
a:</b> 24 giugno 2011 20:35:13 GMT+02:00<br><b>A:</b> Working Group Chairs &=
lt;<a href=3D"mailto:wgchairs@ietf.org">wgchairs@ietf.org</a>&gt;<br><b>Ogge=
tto:</b> <b>Please help the Nomcom </b><br><br></div></blockquote><div></div=
><blockquote type=3D"cite"><div><span>Hi WG chairs,</span><br><span> &nbsp;W=
e have had a good response to the first call for volunteers but the </span><=
br><span>rate at which new volunteers are coming in is slowing down. The Nom=
com </span><br><span>process is best served by a large pool of volunteers dr=
awn from a wide </span><br><span>spectrum of IETF attendees. Where else woul=
d we find this wide spectrum</span><br><span>if not in the WG mailing lists.=
</span><br><span></span><br><span>I would really appreciate it if you can fo=
rward the message onto your</span><br><span>working group mailing lists. </s=
pan><br><span></span><br><span>The latest volunteer status and the second ca=
ll for volunteers can be</span><br><span>found at</span><br><span><a href=3D=
"https://datatracker.ietf.org/ann/nomcom/2964/">https://datatracker.ietf.org=
/ann/nomcom/2964/</a></span><br><span></span><br><span>Thanks in advance for=
 your help.</span><br><span></span><br><span>Suresh Krishnan</span><br><span=
>Nomcom Chair 2011-2012</span><br><span>Email: <a href=3D"mailto:nomcom-chai=
r@ietf.org">nomcom-chair@ietf.org</a>, <a href=3D"mailto:suresh.krishnan@eri=
csson.com"><a href=3D"mailto:suresh.krishnan@ericsson.com">suresh.krishnan@e=
ricsson.com</a></a></span><br><span></span><br></div></blockquote></body></h=
tml>=

--Apple-Mail-6--66304449--

From rifatyu@avaya.com  Sat Jun 25 06:19:29 2011
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From: "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>
To: "splices@ietf.org" <splices@ietf.org>
Date: Sat, 25 Jun 2011 09:14:40 -0400
Thread-Topic: [splices] FW: New Version Notification for draft-loreto-splices-disaggregated-media-02.txt
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Hi,

We have just submitted a new version for the draft-loreto-splices-disaggreg=
ated-media draft.
https://datatracker.ietf.org/doc/draft-loreto-splices-disaggregated-media/

We would appreciate it if you review it and provide us with your feedback.

Regards,
 Rifaat

________________________________________
From: internet-drafts@ietf.org [internet-drafts@ietf.org]
Sent: Saturday, June 25, 2011 9:11 AM
To: Shekh-Yusef, Rifaat (Rifaat)
Cc: salvatore.loreto@ericsson.com; Shekh-Yusef, Rifaat (Rifaat); gonzalo.ca=
marillo@ericsson.com
Subject: New Version Notification for   draft-loreto-splices-disaggregated-=
media-02.txt

A new version of I-D, draft-loreto-splices-disaggregated-media-02.txt has b=
een successfully submitted by Rifaat Shekh-Yusef and posted to the IETF rep=
ository.

Filename:        draft-loreto-splices-disaggregated-media
Revision:        02
Title:           Disaggregated Media in the Session Initiation Protocol (SI=
P)
Creation date:   2011-06-25
WG ID:           Individual Submission
Number of pages: 18

Abstract:
   Disaggregated media refers to the ability for a user to create a
   multimedia session combining different media streams, coming from
   different devices under his or her control, so that they are treated
   by the far end of the session as a single media session.  This
   document lists several use cases that involve disaggregated media in
   SIP.  Additionally, this document analyzes what types of
   disaggregated media can be implemented using existing protocol
   mechanisms, and the pros and cons of using each of those mechanisms.
   Finally, this document describes scenarios that are not covered by
   current mechanisms and proposes new IETF work to cover them.




The IETF Secretariat=

From musgravepj@gmail.com  Tue Jun 28 15:17:35 2011
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Date: Tue, 28 Jun 2011 18:17:30 -0400
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From: Peter Musgrave <musgravepj@gmail.com>
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--00151747598ca5935f04a6cd08ef
Content-Type: text/plain; charset=ISO-8859-1

HI Rifaat,

I like the additions to the draft - makes it very concrete.

This matches what I remember being discussed on the list. It seems that an
unstated principle in section 4 is that the device which will establish the
audio connection is the device which answers the call, even if it is not in
the controller role. Is it worthwhile making that principle explicit?

Do you envisage a case where the device taking only video is the one to
answer the call?

Regards,

Peter Musgrave

On Sat, Jun 25, 2011 at 9:14 AM, Shekh-Yusef, Rifaat (Rifaat) <
rifatyu@avaya.com> wrote:

> Hi,
>
> We have just submitted a new version for the
> draft-loreto-splices-disaggregated-media draft.
> https://datatracker.ietf.org/doc/draft-loreto-splices-disaggregated-media/
>
> We would appreciate it if you review it and provide us with your feedback.
>
> Regards,
>  Rifaat
>
> ________________________________________
> From: internet-drafts@ietf.org [internet-drafts@ietf.org]
> Sent: Saturday, June 25, 2011 9:11 AM
> To: Shekh-Yusef, Rifaat (Rifaat)
> Cc: salvatore.loreto@ericsson.com; Shekh-Yusef, Rifaat (Rifaat);
> gonzalo.camarillo@ericsson.com
> Subject: New Version Notification for
> draft-loreto-splices-disaggregated-media-02.txt
>
> A new version of I-D, draft-loreto-splices-disaggregated-media-02.txt has
> been successfully submitted by Rifaat Shekh-Yusef and posted to the IETF
> repository.
>
> Filename:        draft-loreto-splices-disaggregated-media
> Revision:        02
> Title:           Disaggregated Media in the Session Initiation Protocol
> (SIP)
> Creation date:   2011-06-25
> WG ID:           Individual Submission
> Number of pages: 18
>
> Abstract:
>   Disaggregated media refers to the ability for a user to create a
>   multimedia session combining different media streams, coming from
>   different devices under his or her control, so that they are treated
>   by the far end of the session as a single media session.  This
>   document lists several use cases that involve disaggregated media in
>   SIP.  Additionally, this document analyzes what types of
>   disaggregated media can be implemented using existing protocol
>   mechanisms, and the pros and cons of using each of those mechanisms.
>   Finally, this document describes scenarios that are not covered by
>   current mechanisms and proposes new IETF work to cover them.
>
>
>
>
> The IETF Secretariat
> _______________________________________________
> splices mailing list
> splices@ietf.org
> https://www.ietf.org/mailman/listinfo/splices
>

--00151747598ca5935f04a6cd08ef
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

HI Rifaat,=A0<div><br></div><div>I like the additions to the draft - makes =
it very concrete.=A0<br><div><br></div><div>This matches what I remember be=
ing discussed on the list. It seems that an unstated principle in section 4=
 is that the device which will establish the audio connection is the device=
 which answers the call, even if it is not in the controller role. Is it wo=
rthwhile making that principle explicit?</div>
<div><br></div><div>Do you envisage a case where the device taking only vid=
eo is the one to answer the call?</div><div><br></div><div>Regards,</div><d=
iv><br></div><div>Peter Musgrave<br><br><div class=3D"gmail_quote">On Sat, =
Jun 25, 2011 at 9:14 AM, Shekh-Yusef, Rifaat (Rifaat) <span dir=3D"ltr">&lt=
;<a href=3D"mailto:rifatyu@avaya.com">rifatyu@avaya.com</a>&gt;</span> wrot=
e:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">Hi,<br>
<br>
We have just submitted a new version for the draft-loreto-splices-disaggreg=
ated-media draft.<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-loreto-splices-disaggrega=
ted-media/" target=3D"_blank">https://datatracker.ietf.org/doc/draft-loreto=
-splices-disaggregated-media/</a><br>
<br>
We would appreciate it if you review it and provide us with your feedback.<=
br>
<br>
Regards,<br>
=A0Rifaat<br>
<br>
________________________________________<br>
From: <a href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org<=
/a> [<a href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</=
a>]<br>
Sent: Saturday, June 25, 2011 9:11 AM<br>
To: Shekh-Yusef, Rifaat (Rifaat)<br>
Cc: <a href=3D"mailto:salvatore.loreto@ericsson.com">salvatore.loreto@erics=
son.com</a>; Shekh-Yusef, Rifaat (Rifaat); <a href=3D"mailto:gonzalo.camari=
llo@ericsson.com">gonzalo.camarillo@ericsson.com</a><br>
Subject: New Version Notification for =A0 draft-loreto-splices-disaggregate=
d-media-02.txt<br>
<br>
A new version of I-D, draft-loreto-splices-disaggregated-media-02.txt has b=
een successfully submitted by Rifaat Shekh-Yusef and posted to the IETF rep=
ository.<br>
<br>
Filename: =A0 =A0 =A0 =A0draft-loreto-splices-disaggregated-media<br>
Revision: =A0 =A0 =A0 =A002<br>
Title: =A0 =A0 =A0 =A0 =A0 Disaggregated Media in the Session Initiation Pr=
otocol (SIP)<br>
Creation date: =A0 2011-06-25<br>
WG ID: =A0 =A0 =A0 =A0 =A0 Individual Submission<br>
Number of pages: 18<br>
<br>
Abstract:<br>
 =A0 Disaggregated media refers to the ability for a user to create a<br>
 =A0 multimedia session combining different media streams, coming from<br>
 =A0 different devices under his or her control, so that they are treated<b=
r>
 =A0 by the far end of the session as a single media session. =A0This<br>
 =A0 document lists several use cases that involve disaggregated media in<b=
r>
 =A0 SIP. =A0Additionally, this document analyzes what types of<br>
 =A0 disaggregated media can be implemented using existing protocol<br>
 =A0 mechanisms, and the pros and cons of using each of those mechanisms.<b=
r>
 =A0 Finally, this document describes scenarios that are not covered by<br>
 =A0 current mechanisms and proposes new IETF work to cover them.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat<br>
_______________________________________________<br>
splices mailing list<br>
<a href=3D"mailto:splices@ietf.org">splices@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/splices" target=3D"_blank"=
>https://www.ietf.org/mailman/listinfo/splices</a><br>
</blockquote></div><br></div></div>

--00151747598ca5935f04a6cd08ef--

From wwwrun@rfc-editor.org  Tue Jun 28 15:35:21 2011
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Date: Tue, 28 Jun 2011 15:35:20 -0700 (PDT)
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Subject: [splices] RFC 6271 on Requirements for SIP-Based Session Peering
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 6271

        Title:      Requirements for SIP-Based Session Peering 
        Author:     J-F. Mule
        Status:     Informational
        Stream:     IETF
        Date:       June 2011
        Mailbox:    jf.mule@cablelabs.com
        Pages:      23
        Characters: 54492
        Updates/Obsoletes/SeeAlso:   None

        I-D Tag:    draft-ietf-speermint-requirements-11.txt

        URL:        http://www.rfc-editor.org/rfc/rfc6271.txt

This memo captures protocol requirements to enable session peering of
voice, presence, instant messaging, and other types of multimedia
traffic.  This informational document is intended to link the various
use cases described for session peering to protocol solutions.  This 
document is not an Internet Standards Track specification; it is
published for informational purposes.

This document is a product of the Session PEERing for Multimedia INTerconnect Working Group of the IETF.


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

This announcement is sent to the IETF-Announce and rfc-dist lists.
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The RFC Editor Team
Association Management Solutions, LLC



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From: "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>
To: Peter Musgrave <musgravepj@gmail.com>
Date: Tue, 28 Jun 2011 21:57:30 -0400
Thread-Topic: [splices] FW: New Version Notification for draft-loreto-splices-disaggregated-media-02.txt
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Thanks Peter,

The mechanism is flexable enough to allow any party to answe or initiate a =
call and take the controller role.
Take a look at the following email with an example of the video device init=
iating a call and taking the controller role.
http://www.ietf.org/mail-archive/web/splices/current/msg00123.html

Regards,
 Rifaat




________________________________
From: Peter Musgrave [musgravepj@gmail.com]
Sent: Tuesday, June 28, 2011 6:17 PM
To: Shekh-Yusef, Rifaat (Rifaat)
Cc: splices@ietf.org
Subject: Re: [splices] FW: New Version Notification for draft-loreto-splice=
s-disaggregated-media-02.txt

HI Rifaat,

I like the additions to the draft - makes it very concrete.

This matches what I remember being discussed on the list. It seems that an =
unstated principle in section 4 is that the device which will establish the=
 audio connection is the device which answers the call, even if it is not i=
n the controller role. Is it worthwhile making that principle explicit?

Do you envisage a case where the device taking only video is the one to ans=
wer the call?

Regards,

Peter Musgrave

On Sat, Jun 25, 2011 at 9:14 AM, Shekh-Yusef, Rifaat (Rifaat) <rifatyu@avay=
a.com<mailto:rifatyu@avaya.com>> wrote:
Hi,

We have just submitted a new version for the draft-loreto-splices-disaggreg=
ated-media draft.
https://datatracker.ietf.org/doc/draft-loreto-splices-disaggregated-media/

We would appreciate it if you review it and provide us with your feedback.

Regards,
 Rifaat

________________________________________
From: internet-drafts@ietf.org<mailto:internet-drafts@ietf.org> [internet-d=
rafts@ietf.org<mailto:internet-drafts@ietf.org>]
Sent: Saturday, June 25, 2011 9:11 AM
To: Shekh-Yusef, Rifaat (Rifaat)
Cc: salvatore.loreto@ericsson.com<mailto:salvatore.loreto@ericsson.com>; Sh=
ekh-Yusef, Rifaat (Rifaat); gonzalo.camarillo@ericsson.com<mailto:gonzalo.c=
amarillo@ericsson.com>
Subject: New Version Notification for   draft-loreto-splices-disaggregated-=
media-02.txt

A new version of I-D, draft-loreto-splices-disaggregated-media-02.txt has b=
een successfully submitted by Rifaat Shekh-Yusef and posted to the IETF rep=
ository.

Filename:        draft-loreto-splices-disaggregated-media
Revision:        02
Title:           Disaggregated Media in the Session Initiation Protocol (SI=
P)
Creation date:   2011-06-25
WG ID:           Individual Submission
Number of pages: 18

Abstract:
  Disaggregated media refers to the ability for a user to create a
  multimedia session combining different media streams, coming from
  different devices under his or her control, so that they are treated
  by the far end of the session as a single media session.  This
  document lists several use cases that involve disaggregated media in
  SIP.  Additionally, this document analyzes what types of
  disaggregated media can be implemented using existing protocol
  mechanisms, and the pros and cons of using each of those mechanisms.
  Finally, this document describes scenarios that are not covered by
  current mechanisms and proposes new IETF work to cover them.




The IETF Secretariat
_______________________________________________
splices mailing list
splices@ietf.org<mailto:splices@ietf.org>
https://www.ietf.org/mailman/listinfo/splices


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<div style=3D"FONT-FAMILY: Tahoma; DIRECTION: ltr; COLOR: #000000; FONT-SIZ=
E: 13px">
<div>Thanks Peter,</div>
<div><font size=3D"2" face=3D"tahoma"></font>&nbsp;</div>
<div><font size=3D"2" face=3D"tahoma">The mechanism is flexable enough to a=
llow any party to answe or initiate a call and take the controller role.</f=
ont></div>
<div><font size=3D"2" face=3D"tahoma">Take a look at the following email wi=
th an example of the video device initiating a call and taking the controll=
er role.</font></div>
<div><a href=3D"http://www.ietf.org/mail-archive/web/splices/current/msg001=
23.html">http://www.ietf.org/mail-archive/web/splices/current/msg00123.html=
</a></div>
<div><font size=3D"2" face=3D"tahoma"></font>&nbsp;</div>
<div><font size=3D"2" face=3D"tahoma">Regards,</font></div>
<div><font size=3D"2" face=3D"tahoma">&nbsp;Rifaat</font></div>
<div><font size=3D"2" face=3D"tahoma"></font>&nbsp;</div>
<div><font size=3D"2" face=3D"tahoma"></font>&nbsp;</div>
<div><font size=3D"2" face=3D"tahoma"></font>&nbsp;</div>
<div dir=3D"ltr"><font color=3D"#000000" size=3D"2" face=3D"Tahoma"></font>=
&nbsp;</div>
<div style=3D"DIRECTION: ltr" id=3D"divRpF111305">
<hr tabindex=3D"-1">
<font color=3D"#000000" size=3D"2" face=3D"Tahoma"><b>From:</b> Peter Musgr=
ave [musgravepj@gmail.com]<br>
<b>Sent:</b> Tuesday, June 28, 2011 6:17 PM<br>
<b>To:</b> Shekh-Yusef, Rifaat (Rifaat)<br>
<b>Cc:</b> splices@ietf.org<br>
<b>Subject:</b> Re: [splices] FW: New Version Notification for draft-loreto=
-splices-disaggregated-media-02.txt<br>
</font><br>
</div>
<div></div>
<div>HI Rifaat,&nbsp;
<div><br>
</div>
<div>I like the additions to the draft - makes it very concrete.&nbsp;<br>
<div><br>
</div>
<div>This matches what I remember being discussed on the list. It seems tha=
t an unstated principle in section 4 is that the device which will establis=
h the audio connection is the device which answers the call, even if it is =
not in the controller role. Is it
 worthwhile making that principle explicit?</div>
<div><br>
</div>
<div>Do you envisage a case where the device taking only video is the one t=
o answer the call?</div>
<div><br>
</div>
<div>Regards,</div>
<div><br>
</div>
<div>Peter Musgrave<br>
<br>
<div class=3D"gmail_quote">On Sat, Jun 25, 2011 at 9:14 AM, Shekh-Yusef, Ri=
faat (Rifaat)
<span dir=3D"ltr">&lt;<a href=3D"mailto:rifatyu@avaya.com">rifatyu@avaya.co=
m</a>&gt;</span> wrote:<br>
<blockquote style=3D"BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex=
; PADDING-LEFT: 1ex" class=3D"gmail_quote">
Hi,<br>
<br>
We have just submitted a new version for the draft-loreto-splices-disaggreg=
ated-media draft.<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-loreto-splices-disaggrega=
ted-media/" target=3D"_blank">https://datatracker.ietf.org/doc/draft-loreto=
-splices-disaggregated-media/</a><br>
<br>
We would appreciate it if you review it and provide us with your feedback.<=
br>
<br>
Regards,<br>
&nbsp;Rifaat<br>
<br>
________________________________________<br>
From: <a href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org<=
/a> [<a href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</=
a>]<br>
Sent: Saturday, June 25, 2011 9:11 AM<br>
To: Shekh-Yusef, Rifaat (Rifaat)<br>
Cc: <a href=3D"mailto:salvatore.loreto@ericsson.com">salvatore.loreto@erics=
son.com</a>; Shekh-Yusef, Rifaat (Rifaat);
<a href=3D"mailto:gonzalo.camarillo@ericsson.com">gonzalo.camarillo@ericsso=
n.com</a><br>
Subject: New Version Notification for &nbsp; draft-loreto-splices-disaggreg=
ated-media-02.txt<br>
<br>
A new version of I-D, draft-loreto-splices-disaggregated-media-02.txt has b=
een successfully submitted by Rifaat Shekh-Yusef and posted to the IETF rep=
ository.<br>
<br>
Filename: &nbsp; &nbsp; &nbsp; &nbsp;draft-loreto-splices-disaggregated-med=
ia<br>
Revision: &nbsp; &nbsp; &nbsp; &nbsp;02<br>
Title: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Disaggregated Media in the Sessio=
n Initiation Protocol (SIP)<br>
Creation date: &nbsp; 2011-06-25<br>
WG ID: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Individual Submission<br>
Number of pages: 18<br>
<br>
Abstract:<br>
&nbsp; Disaggregated media refers to the ability for a user to create a<br>
&nbsp; multimedia session combining different media streams, coming from<br=
>
&nbsp; different devices under his or her control, so that they are treated=
<br>
&nbsp; by the far end of the session as a single media session. &nbsp;This<=
br>
&nbsp; document lists several use cases that involve disaggregated media in=
<br>
&nbsp; SIP. &nbsp;Additionally, this document analyzes what types of<br>
&nbsp; disaggregated media can be implemented using existing protocol<br>
&nbsp; mechanisms, and the pros and cons of using each of those mechanisms.=
<br>
&nbsp; Finally, this document describes scenarios that are not covered by<b=
r>
&nbsp; current mechanisms and proposes new IETF work to cover them.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat<br>
_______________________________________________<br>
splices mailing list<br>
<a href=3D"mailto:splices@ietf.org">splices@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/splices" target=3D"_blank"=
>https://www.ietf.org/mailman/listinfo/splices</a><br>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</div>
</body>
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From musgravepj@gmail.com  Wed Jun 29 03:51:28 2011
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Date: Wed, 29 Jun 2011 06:51:22 -0400
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From: Peter Musgrave <musgravepj@gmail.com>
To: "Shekh-Yusef, Rifaat (Rifaat)" <rifatyu@avaya.com>
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Subject: Re: [splices] FW: New Version Notification for draft-loreto-splices-disaggregated-media-02.txt
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Right. I remember now.

This example is a good candidate for the next version of the draft.

Peter



On Tue, Jun 28, 2011 at 9:57 PM, Shekh-Yusef, Rifaat (Rifaat) <
rifatyu@avaya.com> wrote:

>  Thanks Peter,
>
> The mechanism is flexable enough to allow any party to answe or initiate a
> call and take the controller role.
> Take a look at the following email with an example of the video device
> initiating a call and taking the controller role.
> http://www.ietf.org/mail-archive/web/splices/current/msg00123.html
>
> Regards,
>  Rifaat
>
>
>
>
>  ------------------------------
> *From:* Peter Musgrave [musgravepj@gmail.com]
> *Sent:* Tuesday, June 28, 2011 6:17 PM
>
> *To:* Shekh-Yusef, Rifaat (Rifaat)
> *Cc:* splices@ietf.org
> *Subject:* Re: [splices] FW: New Version Notification for
> draft-loreto-splices-disaggregated-media-02.txt
>
>  HI Rifaat,
>
>  I like the additions to the draft - makes it very concrete.
>
>  This matches what I remember being discussed on the list. It seems that
> an unstated principle in section 4 is that the device which will establish
> the audio connection is the device which answers the call, even if it is not
> in the controller role. Is it worthwhile making that principle explicit?
>
>  Do you envisage a case where the device taking only video is the one to
> answer the call?
>
>  Regards,
>
>  Peter Musgrave
>
> On Sat, Jun 25, 2011 at 9:14 AM, Shekh-Yusef, Rifaat (Rifaat) <
> rifatyu@avaya.com> wrote:
>
>> Hi,
>>
>> We have just submitted a new version for the
>> draft-loreto-splices-disaggregated-media draft.
>> https://datatracker.ietf.org/doc/draft-loreto-splices-disaggregated-media/
>>
>> We would appreciate it if you review it and provide us with your feedback.
>>
>> Regards,
>>  Rifaat
>>
>> ________________________________________
>> From: internet-drafts@ietf.org [internet-drafts@ietf.org]
>> Sent: Saturday, June 25, 2011 9:11 AM
>> To: Shekh-Yusef, Rifaat (Rifaat)
>> Cc: salvatore.loreto@ericsson.com; Shekh-Yusef, Rifaat (Rifaat);
>> gonzalo.camarillo@ericsson.com
>> Subject: New Version Notification for
>> draft-loreto-splices-disaggregated-media-02.txt
>>
>> A new version of I-D, draft-loreto-splices-disaggregated-media-02.txt has
>> been successfully submitted by Rifaat Shekh-Yusef and posted to the IETF
>> repository.
>>
>> Filename:        draft-loreto-splices-disaggregated-media
>> Revision:        02
>> Title:           Disaggregated Media in the Session Initiation Protocol
>> (SIP)
>> Creation date:   2011-06-25
>> WG ID:           Individual Submission
>> Number of pages: 18
>>
>> Abstract:
>>   Disaggregated media refers to the ability for a user to create a
>>   multimedia session combining different media streams, coming from
>>   different devices under his or her control, so that they are treated
>>   by the far end of the session as a single media session.  This
>>   document lists several use cases that involve disaggregated media in
>>   SIP.  Additionally, this document analyzes what types of
>>   disaggregated media can be implemented using existing protocol
>>   mechanisms, and the pros and cons of using each of those mechanisms.
>>   Finally, this document describes scenarios that are not covered by
>>   current mechanisms and proposes new IETF work to cover them.
>>
>>
>>
>>
>> The IETF Secretariat
>> _______________________________________________
>> splices mailing list
>> splices@ietf.org
>> https://www.ietf.org/mailman/listinfo/splices
>>
>
>

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Right. I remember now.=A0<div><br></div><div>This example is a good candida=
te for the next version of the draft.</div><div><br></div><div>Peter<br><di=
v><br></div><div><br><br><div class=3D"gmail_quote">On Tue, Jun 28, 2011 at=
 9:57 PM, Shekh-Yusef, Rifaat (Rifaat) <span dir=3D"ltr">&lt;<a href=3D"mai=
lto:rifatyu@avaya.com">rifatyu@avaya.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">




<div>
<div style=3D"font-family:Tahoma;direction:ltr;color:#000000;font-size:13px=
">
<div>Thanks Peter,</div>
<div><font size=3D"2" face=3D"tahoma"></font>=A0</div>
<div><font size=3D"2" face=3D"tahoma">The mechanism is flexable enough to a=
llow any party to answe or initiate a call and take the controller role.</f=
ont></div>
<div><font size=3D"2" face=3D"tahoma">Take a look at the following email wi=
th an example of the video device initiating a call and taking the controll=
er role.</font></div>
<div><a href=3D"http://www.ietf.org/mail-archive/web/splices/current/msg001=
23.html" target=3D"_blank">http://www.ietf.org/mail-archive/web/splices/cur=
rent/msg00123.html</a></div>
<div><font size=3D"2" face=3D"tahoma"></font>=A0</div>
<div><font size=3D"2" face=3D"tahoma">Regards,</font></div>
<div><font size=3D"2" face=3D"tahoma">=A0Rifaat</font></div>
<div><font size=3D"2" face=3D"tahoma"></font>=A0</div>
<div><font size=3D"2" face=3D"tahoma"></font>=A0</div>
<div><font size=3D"2" face=3D"tahoma"></font>=A0</div>
<div dir=3D"ltr"><font color=3D"#000000" size=3D"2" face=3D"Tahoma"></font>=
=A0</div>
<div style=3D"direction:ltr">
<hr>
<font color=3D"#000000" size=3D"2" face=3D"Tahoma"><b>From:</b> Peter Musgr=
ave [<a href=3D"mailto:musgravepj@gmail.com" target=3D"_blank">musgravepj@g=
mail.com</a>]<br>
<b>Sent:</b> Tuesday, June 28, 2011 6:17 PM<div class=3D"im"><br>
<b>To:</b> Shekh-Yusef, Rifaat (Rifaat)<br>
</div><b>Cc:</b> <a href=3D"mailto:splices@ietf.org" target=3D"_blank">spli=
ces@ietf.org</a><br>
<b>Subject:</b> Re: [splices] FW: New Version Notification for draft-loreto=
-splices-disaggregated-media-02.txt<br>
</font><br>
</div><div><div></div><div class=3D"h5">
<div></div>
<div>HI Rifaat,=A0
<div><br>
</div>
<div>I like the additions to the draft - makes it very concrete.=A0<br>
<div><br>
</div>
<div>This matches what I remember being discussed on the list. It seems tha=
t an unstated principle in section 4 is that the device which will establis=
h the audio connection is the device which answers the call, even if it is =
not in the controller role. Is it
 worthwhile making that principle explicit?</div>
<div><br>
</div>
<div>Do you envisage a case where the device taking only video is the one t=
o answer the call?</div>
<div><br>
</div>
<div>Regards,</div>
<div><br>
</div>
<div>Peter Musgrave<br>
<br>
<div class=3D"gmail_quote">On Sat, Jun 25, 2011 at 9:14 AM, Shekh-Yusef, Ri=
faat (Rifaat)
<span dir=3D"ltr">&lt;<a href=3D"mailto:rifatyu@avaya.com" target=3D"_blank=
">rifatyu@avaya.com</a>&gt;</span> wrote:<br>
<blockquote style=3D"border-left:#ccc 1px solid;margin:0px 0px 0px 0.8ex;pa=
dding-left:1ex" class=3D"gmail_quote">
Hi,<br>
<br>
We have just submitted a new version for the draft-loreto-splices-disaggreg=
ated-media draft.<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-loreto-splices-disaggrega=
ted-media/" target=3D"_blank">https://datatracker.ietf.org/doc/draft-loreto=
-splices-disaggregated-media/</a><br>
<br>
We would appreciate it if you review it and provide us with your feedback.<=
br>
<br>
Regards,<br>
=A0Rifaat<br>
<br>
________________________________________<br>
From: <a href=3D"mailto:internet-drafts@ietf.org" target=3D"_blank">interne=
t-drafts@ietf.org</a> [<a href=3D"mailto:internet-drafts@ietf.org" target=
=3D"_blank">internet-drafts@ietf.org</a>]<br>
Sent: Saturday, June 25, 2011 9:11 AM<br>
To: Shekh-Yusef, Rifaat (Rifaat)<br>
Cc: <a href=3D"mailto:salvatore.loreto@ericsson.com" target=3D"_blank">salv=
atore.loreto@ericsson.com</a>; Shekh-Yusef, Rifaat (Rifaat);
<a href=3D"mailto:gonzalo.camarillo@ericsson.com" target=3D"_blank">gonzalo=
.camarillo@ericsson.com</a><br>
Subject: New Version Notification for =A0 draft-loreto-splices-disaggregate=
d-media-02.txt<br>
<br>
A new version of I-D, draft-loreto-splices-disaggregated-media-02.txt has b=
een successfully submitted by Rifaat Shekh-Yusef and posted to the IETF rep=
ository.<br>
<br>
Filename: =A0 =A0 =A0 =A0draft-loreto-splices-disaggregated-media<br>
Revision: =A0 =A0 =A0 =A002<br>
Title: =A0 =A0 =A0 =A0 =A0 Disaggregated Media in the Session Initiation Pr=
otocol (SIP)<br>
Creation date: =A0 2011-06-25<br>
WG ID: =A0 =A0 =A0 =A0 =A0 Individual Submission<br>
Number of pages: 18<br>
<br>
Abstract:<br>
=A0 Disaggregated media refers to the ability for a user to create a<br>
=A0 multimedia session combining different media streams, coming from<br>
=A0 different devices under his or her control, so that they are treated<br=
>
=A0 by the far end of the session as a single media session. =A0This<br>
=A0 document lists several use cases that involve disaggregated media in<br=
>
=A0 SIP. =A0Additionally, this document analyzes what types of<br>
=A0 disaggregated media can be implemented using existing protocol<br>
=A0 mechanisms, and the pros and cons of using each of those mechanisms.<br=
>
=A0 Finally, this document describes scenarios that are not covered by<br>
=A0 current mechanisms and proposes new IETF work to cover them.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat<br>
_______________________________________________<br>
splices mailing list<br>
<a href=3D"mailto:splices@ietf.org" target=3D"_blank">splices@ietf.org</a><=
br>
<a href=3D"https://www.ietf.org/mailman/listinfo/splices" target=3D"_blank"=
>https://www.ietf.org/mailman/listinfo/splices</a><br>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</div></div></div>
</div>

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

--001517475434ae823104a6d79011--

From spromano@unina.it  Wed Jun 29 04:15:26 2011
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Date: Wed, 29 Jun 2011 13:15:17 +0200
From: Simon Pietro Romano <spromano@unina.it>
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Subject: [splices] Provisional agenda for SPLICES
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Hi folks,

as you know, I have requested agenda time (1-hour slot) for a 
face-to-face meeting in Quebec City.
Currently, our agenda time is a s follows:

SPLICES Session 1 (1 hour)
Friday, Afternoon Session II 1415-1515
Room Name: 2101

I'm gonna put on the agenda the following two documents:

draft-loreto-splices-disaggregated-media-02.txt
draft-yusef-splices-invoke-01.txt

If you have any further request, please drop me an e-mail.

Cheers,

Simon

-- 
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    ~~~~~~~~~~~~~~~~~~~~~~o00~~(_)~~00o~~~~~~~~~~~~~~~~~~~~~~~~
                     Simon Pietro Romano
               Universita' di Napoli Federico II
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         Phone: +39 081 7683823 -- Fax: +39 081 7684219
                 e-mail: spromano@unina.it
           http://www.comics.unina.it/simonpietro.romano

     <<Molti mi dicono che lo scoraggiamento è l'alibi degli
    idioti. Ci rifletto un istante; e mi scoraggio>>. Magritte.
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From gonzalo.camarillo@ericsson.com  Wed Jun 29 23:06:50 2011
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Subject: Re: [splices] Using Two Separate Devices to Start a Conversation proposal
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Hi Paul,

yes, Join was considered at that point. As you say, the semantics are
similar to Join but not identical (specially with respect of how things
are presented to the user). Overloading Join may not be the best way to
proceed.

If you are concerned with supporting another primitive in addition to
Join, the code base for Join and the new primitive would be almost
identical.

Cheers,

Gonzalo

On 21/06/2011 3:32 PM, Paul Kyzivat wrote:
> IMO there is nothing wrong with using Join to bring together one dialog 
> with audio and another with video. I don't see why we would create 
> another mechanism when that one is there to be used. Certainly it is a 
> degenerate case of Join, but there is nothing wrong with degenerate cases.
> 
> Of course there might be UAs that are capable of supporting the case 
> where the media in the dialogs is disjoint, but not the case where there 
> is common media in the dialogs that must be mixed. But surely we can 
> find an error code to cover that case.
> 
> The key issue here is what UA we expect to support this capability. IMO 
> a user that wants to "splice" some of its devices together should take 
> the responsibility of finding a UA that can handle the coordination, 
> rather than depending upon the far end to do it.
> 
> 	Thanks,
> 	Paul
> 
> On 6/21/2011 4:09 AM, Gonzalo Camarillo wrote:
>> Hi Alan,
>>
>>>> As Dale mentioned, this also depends on the other end doing the "right
>>>> thing" with this case. It is a leap of faith to assume it will realize it
>>>> should accept both streams and use one for input and the other for output.
>>>>
>>>> ISTM its at least as likely that it would accept the first stream, reject
>>>> the second, and then have only a one-way "conversation".
>>>
>>> OK, I understand the problem right now.  The problem is that this is a
>>> special kind of "Join" operation.  It is kind of a join, but not
>>> quite.
>>>
>>> The operation we are after is perhaps similar to 3.3.8 Far-Fork in RFC 5850:
>>>
>>> I suspect instead of Join, we need a new primitive, lets call it
>>> Splice or Merge which requests the UAS to add the media in the INVITE
>>> into the dialog that is referenced.
>>
>> Yes... this is exactly what we proposed 5 years ago in one of the drafts
>> that, eventually, led to the creation of the SPLICES WG:
>>
>> http://tools.ietf.org/html/draft-loreto-sipping-dialog-correlation-01
>>
>> The idea was that the media streams of a given session could be
>> established using more than one SIP dialog.
>>
>> Note that most SPLICES-related discussions up to a couple of months ago
>> were based on the fact that such a mechanism (i.e., using different
>> dialogs to establish a single session) would require support from the
>> remote end point. That is also the point Paul made in his comment above.
>> I personally would be OK with a mechanism along these lines, though.
>>
>> Cheers,
>>
>> Gonzalo
>>
>>


From dworley@avaya.com  Thu Jun 30 15:06:43 2011
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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>, Alan Johnston <alan.b.johnston@gmail.com>
Date: Thu, 30 Jun 2011 18:06:38 -0400
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>> As Dale mentioned, this also depends on the other end doing the "right
>> thing" with this case. It is a leap of faith to assume it will realize i=
t
>> should accept both streams and use one for input and the other for outpu=
t.
>>
>> ISTM its at least as likely that it would accept the first stream, rejec=
t
>> the second, and then have only a one-way "conversation".
>
> OK, I understand the problem right now.  The problem is that this is a
> special kind of "Join" operation.  It is kind of a join, but not
> quite.
>
> The operation we are after is perhaps similar to 3.3.8 Far-Fork in RFC 58=
50:
>
> I suspect instead of Join, we need a new primitive, lets call it
> Splice or Merge which requests the UAS to add the media in the INVITE
> into the dialog that is referenced.

OK, now I see the problem with using Join.

UA A1 sends an INVITE (requesting only audio) to UA B (which supports
both audio and video).

UA A2 sends an INVITE (requesting only video) to UA B (which supports
both audio and video), with Join to the first dialog.

Then we should expect that UA B will send its audio to A1 and its
video to A2, within the semantics of Join.

The difficulty is that UA B still sees the situation as two different
dialogs.  E.g., if A1 sends a REFER/Refer-To: C to B, the A1-B dialog
will be replaced by the C-B dialog, but the A2-B dialog will be
undisturbed.

Which is not the behavior we want if we are trying to logically bond
A1 and A2 to act as a single UA.

Using Join would have been convenient as it is already defined, and
theoretically, UAs already support it.

Dale
