[AVT] RTP translator or mixer for multi-party real-time text

["Gunnar Hellstrom" <[e-mail]>]

A new version of draft-hellstrom-text-conference is available at
 
<http://www.ietf.org/internet-drafts/draft-hellstrom-text-conference-01.txt>
http://www.ietf.org/internet-drafts/draft-hellstrom-text-conference-01.txt

 
It brings up the question if the RTP translator model or mixer model shall
be specified for multi-party real-time text.
 
In its earlier version, only the translator model was mentioned, using
multiple SSRCs as source identification in one single RTP session. That, in
fact seemed not to require any additional standardisation.
 
However, two problems have been identified with the RTP translator model:
 
1. Some RTP implementations do not implement support for multiple SSRC:s in
the same session. There is a risk that it will cause failures.
 
2. When adding a new stream to the session, a new SSRC is started to be
used. There seems to be a habit in RTP to throw away some initial packets
until it is clearly evident that the new SSRC represents an added source.
This behaviour is very unfavourable for real-time text, where information
can be carried from the first packet.
 
Because of these problems, the RTP mixer model has been briefly  introduced
as an alternative, requireing an in-line source identification in the text
medium.
 
It is of course not favourable to have two alternatives for one task.
Therefore I ask: were the comments on the translator model valid, and can
anything be done about them, or shall we move to only specifying the mixer
model?
 
Info on the draft:
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Title : Text media handling in RTP based real-time conferences 

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

Filename : draft-hellstrom-text-conference-01.txt

Pages : 9

Date : 2009-07-12

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. An alternative method using a single RTP
stream and source identification inline in the text stream is also
described.

A URL for this Internet-Draft is:

 
<http://www.ietf.org/internet-drafts/draft-hellstrom-text-conference-01.txt>
http://www.ietf.org/internet-drafts/draft-hellstrom-text-conference-01.txt

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/Gunnar
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Gunnar Hellström
Omnitor
[e-mail]
Tel: [phone]
www.omnitor.se <http://www.omnitor.se/>