Browse Prior Art Database

RTP Payload Format for PureVoice(tm) Audio (RFC2658)

IP.com Disclosure Number: IPCOM000003247D
Original Publication Date: 1999-Aug-01
Included in the Prior Art Database: 2000-Sep-13
Document File: 8 page(s) / 20K

Publishing Venue

Internet Society Requests For Comment (RFCs)

Related People

K. McKay: AUTHOR

Abstract

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 text was extracted from a ASCII document.
This is the abbreviated version, containing approximately 13% of the total text.

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 Notice

Copyright (C) The Internet Society (1999). All Rights Reserved.

ABSTRACT

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.

1 Introduction

This document describes how compressed PureVoice audio as produced by

the Qualcomm PureVoice CODEC [1] 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 [3].

2 Background

The Electronic Industries Association (EIA) & Telecommunications

Industry Association (TIA) standard IS-733 [1] 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

Eudora(r).

The Qcelp CODEC [1] 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...