Text media handling in RTP based real-time conferences

Document Type Replaced Internet-Draft (individual)
Authors Gunnar Hellstrom  , Arnoud Wijk 
Last updated 2011-03-14
Replaced by draft-hellstrom-mmusic-multi-party-rtt
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This Internet-Draft is no longer active. A copy of the expired Internet-Draft can be found at


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.


Gunnar Hellstrom (gunnar.hellstrom@omnitor.se)
Arnoud Wijk (arnoud@realtimetext.org)

(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)