SIP Call Control: Call Handoff
draft-dean-handoff-00
| Document | Type |
Expired Internet-Draft
(individual)
Expired & archived
|
|
|---|---|---|---|
| Authors | Billy Biggs , Rick Dean | ||
| Last updated | 2001-01-22 | ||
| 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This Internet-Draft is no longer active.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 a method of signaling call handoff using SIP. Call handoff is the process of moving one end of an active call without creating a new logical call. Handoff is an important primitive for providing advanced telephony services.
Authors
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)