RTP Payload Format for the iSAC Codec

Document Type Expired Internet-Draft (payload WG)
Authors Tina Grand  , Paul Jones  , Pascal Huart  , Turaj Shabestary , Harald Alvestrand 
Last updated 2015-10-14 (latest revision 2013-02-08)
Replaces draft-legrand-rtp-isac
Stream Internet Engineering Task Force (IETF)
Intended RFC status Proposed Standard
Expired & archived
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Stream WG state In WG Last Call
Document shepherd Roni Even
IESG IESG state Expired (IESG: Dead)
Action Holders
Consensus Boilerplate Unknown
Telechat date
Responsible AD Richard Barnes
IESG note Roni Even (ron.even.tlv@gmail.com) is the Document Shepherd
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This Internet-Draft is no longer active. A copy of the expired Internet-Draft can be found at


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).


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.)