Requirements for interworking between RTC-Web and SIP-RTP protocols
draft-marjou-dispatch-rtcweb-sip-rtp-interwk-reqs-00
| Document | Type |
Expired Internet-Draft
(individual)
Expired & archived
|
|
|---|---|---|---|
| Authors | Xavier Marjou , Jean-Francois Jestin | ||
| Last updated | 2011-02-09 | ||
| 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This Internet-Draft is no longer active.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
In the context of [RTC-Web], some work is emerging to make real-time communications possible in a web browser. This document defines a minimal set of requirements so that such applications interoperate with SIP-RTP applications.
Authors
Xavier Marjou
Jean-Francois Jestin
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)