[dispatch] Text media handling in RTP based real-time conferences

[Arnoud van Wijk <[e-mail]>]

Hi everybody,

Since this draft was submitted in March ago, I like to ask DISPATCH if 
the AVTCORE WG is the best WG for it.
I think so, since this does handle media handling in conferences.
We have edited this draft based on the discussions in AVT core in march 
about SSRC multiplexing.

It is very important that when using RTT streams in a multiparty call 
that it will work with all SIP based call control environments.
I feel that we need together with AVT to work out the details to ensure 
that this remains possible.

Several solutions are currently being implemented in RTT systems by 
several companies and the longer we wait..the harder it gets to ensure 
the best implementation to be used.

So, this is quite important for real-world cases.

I am looking forward to your feedback as always.

sincerely

Arnoud van Wijk



> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>
> 	Title           : Text media handling in RTP based real-time conferences
> 	Author(s)       : G. Hellstrom, A. Wijk
> 	Filename        : draft-hellstrom-text-conference-04.txt
> 	Pages           : 9
> 	Date            : 2011-03-14
>
> 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 single
> stream RTP session, identifying each stream with its own CSRC.
> 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.  Two
> alternative methods using a single RTP stream and source
> identification inline in the text stream are also described.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-hellstrom-text-conference-04.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.
>
> <ftp://ftp.ietf.org/internet-drafts/draft-hellstrom-text-conference-04.txt>
>