Re: I-D Action:draft-loreto-sipping-context-id-requirements-02.txt

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Hi John,

thanks for your comments,

Elwell, John wrote:
<snip>

I agree that handling different media within a session on different
devices on the calling side and/or the called side is an interesting
scenario. However, I believe one of the requirements is as follows:
"REQx: It must be possible for a user participating in a multi-media
session to use different devices for different media, without requiring
any special support at the peer UA, other than the need to support
different remote IP addresses for different media."
yes, you can use 3pcc mechanism to solve all the use cases that is a fact.
Also the session mobility draft, in section 5.4.3 "Transfer to multiple devices"
http://tools.ietf.org/html/draft-shacham-sipping-session-mobility-05#page-22
suggests the usage of 3pcc mechanism "/to overcame this problem/" that
"/there is currently no standard way to associate multiple sessions as part of a single call in SIP./"

And yes the 3pcc hides everything you do on your side to the peer UA.

Of course, using 3pcc make users want to use different device "independent" from what you call "special support"
at the peer multimedia devices.
However there is the implicit requirement that the user has to have among his device, at least one that support 3pcc and is able to behave as controller. Moreover there is an even more strict requirement, the controller
must not leave the call for all the session time.

so lets say you are at your office and join a call using your mobile phone for the audio, and your computer
for the video and the messaging part.
You can do that because the SIP UA on your computer probably support the 3pcc, and then act as controller. However if the call lasts too long and you have to go home, but you want continue to be involved in the call only with the audio you cannot turn off your computer because if you do you will automatically thrown out from the call; and if you leave the computer on, I am not sure you will then be able to hang up completely the
call using only your mobile phone and not the controller.
Otherwise users want to use different devices in this way would be
dependent on special support at peer multimedia devices.
no question that you have to relay on "special support" at the peer, and so maybe legacy phone won't be able to support it;
but special support does not necessarily means complicate to implement.

regards
Sal
John





-----Original Message-----
From: i-d-announce-bounces@xxxxxxxx [mailto:i-d-announce-bounces@xxxxxxxx] On Behalf Of Internet-Drafts@xxxxxxxx
Sent: 06 March 2009 07:15
To: i-d-announce@xxxxxxxx
Subject: I-D Action:draft-loreto-sipping-context-id-requirements-02.txt

A New Internet-Draft is available from the on-line Internet-Drafts directories.

Title : Requirements for Dialog Correlation in the Session Initiation Protocol (SIP)
	Author(s)       : G. Camarillo, S. Loreto
Filename : draft-loreto-sipping-context-id-requirements-02.txt
	Pages           : 10
	Date            : 2009-03-05

This document justifies the need and lists the requirements for
correlating SIP (Session Initiation Protocol) dialogs.  The
correlated dialogs may or may not be related to the same multimedia
session.  Being able to logically correlate multiple SIP dialogs is
useful for applications that, for different reasons, need to
establish several SIP dialogs to provide a given service.  The
logical correlation of two SIP dialogs is also useful, for instance,
to correlate an incoming with an outgoing dialog at a B2BUA.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-loreto-sipping-conte
xt-id-requirements-02.txt

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

Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

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Use sip-implementors@xxxxxxxxxxxxxxx for questions on current sip
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_______________________________________________
Sipping mailing list  https://www.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@xxxxxxxxxxxxxxx for questions on current sip
Use sip@xxxxxxxx for new developments of core SIP

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