Using SIP for Peer-to-Peer Third-Party Call Control
draft-mahy-sip-peer-3pcc-00
| Document | Type |
Expired Internet-Draft
(individual)
Expired & archived
|
|
|---|---|---|---|
| Author | Rohan Mahy | ||
| Last updated | 2000-11-17 | ||
| RFC stream | (None) | ||
| Intended RFC status | (None) | ||
| Formats | |||
| 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This Internet-Draft is no longer active.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
The 3rd party call control draft [2] demonstrates a usage of SIP [3] with some of the enhancements of RFC2543bis [4]. This usage requires that a 3pcc controller remain in the signaling path and maintain state for the duration of the call. While this is necessary in certain situations (ex: protocol translation: H.323 [5] to SIP, SIP to RTSP [6], HTTP [7] to SIP), it is sub-optimal in pure SIP environments. This draft demonstrates a usage of the REFER method [8] and SIP for presence [9] to allow authorized peers to participate in call control.
Authors
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)