Text media handling in RTP based real-time conferences
draft-hellstrom-text-conference-04
Document | Type |
Replaced Internet-Draft
(individual)
Expired & archived
|
|
---|---|---|---|
Authors | Gunnar Hellstrom , Arnoud Wijk | ||
Last updated | 2011-03-14 | ||
Replaced by | draft-hellstrom-mmusic-multi-party-rtt | ||
RFC stream | (None) | ||
Intended RFC status | (None) | ||
Formats | |||
Stream | Stream state | (No stream defined) | |
Consensus boilerplate | Unknown | ||
RFC Editor Note | (None) | ||
IESG | IESG state | Replaced by draft-hellstrom-mmusic-multi-party-rtt | |
Telechat date | (None) | ||
Responsible AD | (None) | ||
Send notices to | (None) |
This Internet-Draft is no longer active. A copy of the expired Internet-Draft is available in these formats:
Abstract
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.
Authors
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)