SIP Call Control Services
draft-ietf-mmusic-sip-cc-01
Document | Type |
Expired Internet-Draft
(mmusic WG)
Expired & archived
|
|
---|---|---|---|
Authors | Jonathan Rosenberg , Henning Schulzrinne | ||
Last updated | 1999-06-25 | ||
RFC stream | Internet Engineering Task Force (IETF) | ||
Intended RFC status | (None) | ||
Formats | |||
Additional resources | Mailing list discussion | ||
Stream | WG state | WG Document | |
Document shepherd | (None) | ||
IESG | IESG state | Expired | |
Consensus boilerplate | Unknown | ||
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This Internet-Draft is no longer active.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 a set of extensions to SIP which allow for various call control services. Example services include blind transfer, transfer with consultation, multi-party calls, bridged conferences, and ad-hoc conferencing. The services are supported in a fully distributed manner, so that they can be provided without a central conference server. However, a SIP proxy can act as a conference server to provide these services. For the various services described here, we overview the requirements for the service, and specify the protocol functions needed to support it. We then define a basic set of SIP primitives which can be used to construct these services, and others.
Authors
Jonathan Rosenberg
Henning Schulzrinne
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)