RTP Payload Format for the Speex Codec
draft-herlein-avt-rtp-speex-00
Document | Type |
Replaced Internet-Draft
(individual)
Expired & archived
|
|
---|---|---|---|
Author | Greg Herlein | ||
Last updated | 2009-02-24 (Latest revision 2004-03-09) | ||
Replaced by | draft-ietf-avt-rtp-speex | ||
RFC stream | (None) | ||
Intended RFC status | (None) | ||
Formats | |||
Stream | Stream state | (No stream defined) | |
Consensus boilerplate | Unknown | ||
RFC Editor Note | (None) | ||
IESG | IESG state | Replaced by draft-ietf-avt-rtp-speex | |
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This Internet-Draft is no longer active.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
Speex is an open-source voice codec suitable for use in Voice over IP (VoIP) type applications. This document describes the payload format for Speex generated bit streams within an RTP packet. Also included here are the necessary details for the use of Speex with the Session Description Protocol (SDP) and a preliminary method of using Speex within H.323 applications.
Authors
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)