SIP Extensions for supporting Distributed Call State
draft-dcsgroup-mmusic-state-00
| Document | Type |
Expired Internet-Draft
(individual)
Expired & archived
|
|
|---|---|---|---|
| Authors | John Pickens , Dr. K. K. Ramakrishnan , Eric Miller , Burcak N. Beser , Bill Marshall , Ed Miller , Glenn Russell | ||
| Last updated | 1999-06-29 | ||
| 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This Internet-Draft is no longer active.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
This document describes extensions to the Session Initiation Protocol RFC(2543) for supporting telephony services using the Distributed Call Signaling architecture described in [2]. This document discusses extensions to the host parameter in the SIP URL for supporting the following: (a) a call signaling architecture where call state is distributed to the clients during call setup and stored there for the duration of the call, (b) support for local number portability and (c) relaxation of the strong coupling between the phone number and its gateway in the telephony subscriber URL as defined in RFC 2543. Additional headers in SIP messages for passing call state information between the clients and their proxies, and support for PSTN-based operator services are also discussed in this document. In addition, the need for the Also and Replaces headers previously discussed in [3]) are mentioned here.
Authors
John Pickens
Dr. K. K. Ramakrishnan
Eric Miller
Burcak N. Beser
Bill Marshall
Ed Miller
Glenn Russell
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)