RTP Payload Format for PureVoice(tm) Audio
RFC 2658

Document Type RFC - Proposed Standard (August 1999; No errata)
Was draft-mckay-qcelp (individual)
Author Kyle McKay 
Last updated 2013-03-02
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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.


   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",
   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

K. McKay                    Standards Track                     [Page 1]
RFC 2658       RTP Payload Format for PureVoice(tm) Audio    August 1999

   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 the following format:

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   |                      RTP Header [2]                           |
   |RR | LLL | NNN |                                               |
   +-+-+-+-+-+-+-+-+       one or more codec data frames           |
   |                             ....                              |

   The RTP header has the expected values as described in [2].  The
   extension bit is not set and this payload type never sets the marker
   bit.  The codec data frames are aligned on octet boundaries.  When
   interleaving is in use and/or multiple codec data frames are present
   in a single RTP packet, the timestamp is, as always, that of the
   oldest data represented in the RTP packet.  The other fields have the
   following meaning:

   Reserved (RR): 2 bits
      MUST be set to zero by sender, SHOULD be ignored by receiver.

   Interleave (LLL): 3 bits
      MUST have a value between 0 and 5 inclusive.  The remaining two
      values (6 and 7) MUST not be used by senders.  If this field is
      non-zero, interleaving is enabled.  All receivers MUST support
      interleaving.  Senders MAY support interleaving.  Senders that do
      not support interleaving MUST set field LLL and NNN to zero.

   Interleave Index (NNN): 3 bits
      MUST have a value less than or equal to the value of LLL.  Values
      of NNN greater than the value of LLL are invalid.

K. McKay                    Standards Track                     [Page 2]
RFC 2658       RTP Payload Format for PureVoice(tm) Audio    August 1999

3.1 Receiving Invalid Values

   On receipt of an RTP packet with an invalid value of the LLL or NNN
   field, the RTP packet MUST be treated as lost by the receiver for the
   purpose of generating erasure frames as described in section 4.

3.2 CODEC data frame format
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