RTP Payload Format for the iSAC Codec
draft-ietf-avt-rtp-isac-04

Document Type Expired Internet-Draft (payload WG)
Last updated 2015-10-14 (latest revision 2013-02-08)
Replaces draft-legrand-rtp-isac
Stream IETF
Intended RFC status Proposed Standard
Formats
Expired & archived
plain text pdf html bibtex
Stream WG state In WG Last Call
Document shepherd Roni Even
IESG IESG state Expired (IESG: Dead)
Consensus Boilerplate Unknown
Telechat date
Responsible AD Richard Barnes
IESG note Roni Even (ron.even.tlv@gmail.com) is the Document Shepherd
Send notices to (None)

This Internet-Draft is no longer active. A copy of the expired Internet-Draft can be found at
https://www.ietf.org/archive/id/draft-ietf-avt-rtp-isac-04.txt

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 Grand
Paul Jones (paulej@packetizer.com)
Pascal Huart (phuart@cisco.com)
Turaj Shabestary (turajs@google.com)
Harald Alvestrand (hta@google.com)

(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)