RTP Payload Format for PureVoice(tm) Audio (RFC2658)
Original Publication Date: 1999-Aug-01
Included in the Prior Art Database: 2000-Sep-13
Internet Society Requests For Comment (RFCs)
This document describes the RTP payload format for PureVoice(tm) Audio. The packet format supports variable interleaving to reduce the effect of packet loss on audio quality.
Network Working Group K. McKay
Request for Comments: 2658 QUALCOMM Incorporated
Category: Standards Track August 1999
RTP Payload Format for PureVoice(tm) Audio
Status of this Memo
This document specifies an Internet standards track protocol for the
Internet community, and requests discussion and suggestions for
improvements. Please refer to the current edition of the "Internet
Official Protocol Standards" (STD 1) for the standardization state
and status of this protocol. Distribution of this memo is unlimited.
Copyright (C) The Internet Society (1999). All Rights Reserved.
This document describes the RTP payload format for PureVoice(tm)
Audio. The packet format supports variable interleaving to reduce
the effect of packet loss on audio quality.
This document describes how compressed PureVoice audio as produced by
the Qualcomm PureVoice CODEC  may be formatted for use as an RTP
payload type. A method is provided to interleave the output of the
compressor to reduce quality degradation due to lost packets.
Furthermore, the sender may choose various interleave settings based
on the importance of low end-to-end delay versus greater tolerance
for lost packets.
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 .
The Electronic Industries Association (EIA) & Telecommunications
Industry Association (TIA) standard IS-733  defines an audio
compression algorithm for use in CDMA applications. In addition to
being the standard CODEC for all wireless CDMA terminals, the
Qualcomm PureVoice CODEC (a.k.a. Qcelp) is used in several Internet
applications most notably JFax(tm), Apple(r) QuickTime(tm), and
The Qcelp CODEC  compresses each 20 milliseconds of 8000 Hz, 16-
bit sampled input speech into one of four different size output
frames: Rate 1 (266 bits), Rate 1/2 (124 bits), Rate 1/4 (54 bits)
or Rate 1/8 (20 bits). The CODEC chooses the output frame rate based
on analysis of the input speech and the current operating mode
(either normal or reduced rate). For typical speech patterns, this
results in an average output of 6.8 k bits/sec for normal mode and
4.7 k bits/sec for reduced rate mode.
3 RTP/Qcelp Packet Format
The RTP timestamp is in 1/8000 of a second units. The RTP payload
data for the Qcelp CODEC has t...