Session Initiation Protocol Call Control -
Conferencing for User Agents
draft-johnston-sipping-cc-conferencing-01
| Document | Type | Expired Internet-Draft (individual) | |
|---|---|---|---|
| Author | Orit Levin | ||
| Last updated | 2003-02-11 | ||
| Stream | (None) | ||
| Intended RFC status | (None) | ||
| Formats |
Expired & archived
plain text
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| 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) |
https://www.ietf.org/archive/id/draft-johnston-sipping-cc-conferencing-01.txt
Abstract
This document defines conferencing call control features for the Session Initiation Protocol (SIP). This document builds on the Conferencing Requirements and Framework documents to define how a tightly coupled SIP conference works. The approach is explored from different user agent (UA) types perspective: conference-unaware, conference-aware and focus UAs. The use of URIs in conferencing, OPTIONS for capabilities discovery, and call control using REFER are covered in detail with example call flow diagrams.
Authors
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)