RTP Payload Format for the iSAC Codec
draft-ietf-avt-rtp-isac-04
Document | Type |
Expired Internet-Draft
(payload WG)
Expired & archived
|
|
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Authors | Tina le Grand , Paul Jones , Pascal Huart , Turaj Zakizadeh Shabestary, Harald T. Alvestrand | ||
Last updated | 2015-10-14 (Latest revision 2013-02-08) | ||
Replaces | draft-legrand-rtp-isac | ||
RFC stream | Internet Engineering Task Force (IETF) | ||
Intended RFC status | Proposed Standard | ||
Formats | |||
Additional resources | Mailing list discussion | ||
Stream | WG state | In WG Last Call | |
Document shepherd | Roni Even | ||
IESG | IESG state | Expired (IESG: Dead) | |
Action Holders |
(None)
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Consensus boilerplate | Unknown | ||
Telechat date | (None) | ||
Responsible AD | Richard Barnes | ||
IESG note | ** No value found for 'doc.notedoc.note' ** | ||
Send notices to | (None) |
This Internet-Draft is no longer active. A copy of the expired Internet-Draft is available in these formats:
Abstract
iSAC is a proprietary wideband speech and audio codec developed by Global IP Solutions (now part of Google), suitable for use in Voice over IP applications. This document describes the payload format for iSAC generated bit streams within a Real-Time Protocol (RTP) packet. Also included here are the necessary details for the use of iSAC with the Session Description Protocol (SDP).
Authors
Tina le Grand
Paul Jones
Pascal Huart
Turaj Zakizadeh Shabestary
Harald T. Alvestrand
(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)