Subject: [AVT] A multi-party text draft for RTP is available from IETF FW: I-D Action:draft-hellstrom-text-conference-00.txt

From: Gunnar Hellström <[email-masked]>
Date: Fri, 4 Jan 2008 00:42:47 +0100

There is a new draft available that I would like to get avt:s attention to.
It is a specification on how to use the real-time text RTP transport RFC
4103 in multi-party sessions.

The spec does not define any new protocol features. Its ambition is just to
create a harmonized view of how to handle multi-party sessions with text.

It makes use of the Translator feature of RTP, described in RFC 3550 section
7.1.

I think this specification is needed because of all references to RFC 4103
appearing will now and then require multi-party sessions. One such example
is the use for emergency services specified in draft-ietf-ecrit-phonebcp. 

Do you find the approach described feasible?
Any comments are welcome.

Gunnar

-----Original Message-----

From: [email-masked] [mailto:[email-masked]
<mailto:[email-masked]> ]

Sent: Thursday, January 03, 2008 10:30 PM

To: [email-masked]

Subject: I-D Action:draft-hellstrom-text-conference-00.txt 

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

Title : Text media handling in RTP based real-time and

message conferences

Author(s) : G. Hellstrom, A. Wijk

Filename : draft-hellstrom-text-conference-00.txt

Pages : 8

Date : 2008-01-03

This memo specifies methods for text media handling in multi-party calls,
where the text is carried by the RTP protocol. Real-time text is carried in
a time-sampled mode according to RFC 4103. Centralized multi-party handling
of real-time text is achieved through a media control unit coordinating
multiple RTP text streams into one RTP session, identifying each stream with
its own SSRC. Identification for the streams are provided through the RTCP
messages. This mechanism enables the receiving application to present the
received real-time text medium in different ways according to user
preferences. Some presentation related features are also described
explaining suitable variations of transmission and presentation of text.

Call control features are described for the SIP environment, while the
transport mechanisms should be suitable for any IP based call control
environment using RTP transport.

A URL for this Internet-Draft is:

http://www.ietf.org/internet-drafts/draft-hellstrom-text-conference-00.txt
<http://www.ietf.org/internet-drafts/draft-hellstrom-text-conference-00.txt>

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

Gunnar Hellström

Omnitor

[email-masked]

Tel: +46708204288

www.omnitor.se <outbind://165/www.omnitor.se> 

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