Real-time text solutions for multi-party sessions
draft-hellstrom-avtcore-multi-party-rtt-solutions-06
| Document | Type |
This is an older version of an Internet-Draft whose latest revision state is "Expired".
Expired & archived
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| Author | Gunnar Hellstrom | ||
| Last updated | 2021-06-20 (Latest revision 2020-12-17) | ||
| Replaces | draft-hellstrom-mmusic-multi-party-rtt | ||
| RFC stream | (None) | ||
| Formats | |||
| Additional resources | |||
| Stream | Stream state | (No stream defined) | |
| Consensus boilerplate | Unknown | ||
| RFC Editor Note | (None) | ||
| IESG | IESG state | Expired | |
| 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This Internet-Draft is no longer active.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 document specifies methods for Real-Time Text (RTT) media handling in multi-party calls. The main discussed transport is to carry Real-Time text by the RTP protocol in a time-sampled mode according to RFC 4103. The mechanisms enable the receiving application to present the received real-time text media, separated per source, 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. A number of alternative methods for providing the multi-party negotiation, transmission and presentation are discussed and a recommendation for the main ones is provided. The main solution for SIP based centralized multi-party handling of real-time text is achieved through a media control unit coordinating multiple RTP text streams into one RTP stream. Alternative methods using a single RTP stream and source identification inline in the text stream are also described, one of them being provided as a lower functionality fallback method for endpoints with no multi-party awareness for RTT. Bridging methods where the text stream is carried without the contents being dealt with in detail by the bridge are also discussed. Brief information is also provided for multi-party RTT in the WebRTC environment. The intention is to provide background for decisions, specification and implementation of selected methods.
Authors
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