Real-time Transport Protocol (RTP) Payload Format for internet Low Bit Rate Codec (iLBC) Speech
RFC 3952

Document Type RFC - Experimental (December 2004; No errata)
Last updated 2013-03-02
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Network Working Group                                           A. Duric
Request for Comments: 3952                                         Telio
Category: Experimental                                       S. Andersen
                                                      Aalborg University
                                                           December 2004

           Real-time Transport Protocol (RTP) Payload Format
             for internet Low Bit Rate Codec (iLBC) Speech

Status of this Memo

   This memo defines an Experimental Protocol for the Internet
   community.  It does not specify an Internet standard of any kind.
   Discussion and suggestions for improvement are requested.
   Distribution of this memo is unlimited.

Copyright Notice

   Copyright (C) The Internet Society (2004).

Abstract

   This document describes the Real-time Transport Protocol (RTP)
   payload format for the internet Low Bit Rate Codec (iLBC) Speech
   developed by Global IP Sound (GIPS).  Also, within the document there
   are included necessary details for the use of iLBC with MIME and
   Session Description Protocol (SDP).

Table of Contents

   1. Introduction. . . . . . . . . . . . . . . . . . . . . . . . . .  2
   2. Background. . . . . . . . . . . . . . . . . . . . . . . . . . .  2
   3. RTP Payload Format. . . . . . . . . . . . . . . . . . . . . . .  3
      3.1. Bitstream definition . . . . . . . . . . . . . . . . . . .  3
      3.2. Multiple iLBC frames in a RTP packet . . . . . . . . . . .  6
   4. IANA Considerations . . . . . . . . . . . . . . . . . . . . . .  7
      4.1. Storage Mode . . . . . . . . . . . . . . . . . . . . . . .  7
      4.2. MIME registration of iLBC. . . . . . . . . . . . . . . . .  8
   5. Mapping to SDP Parameters . . . . . . . . . . . . . . . . . . .  9
   6. Security Considerations . . . . . . . . . . . . . . . . . . . . 11
   7. References. . . . . . . . . . . . . . . . . . . . . . . . . . . 11
      7.1. Normative References . . . . . . . . . . . . . . . . . . . 11
      7.2. Informative References . . . . . . . . . . . . . . . . . . 12
   8. Acknowledgements. . . . . . . . . . . . . . . . . . . . . . . . 12
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 12
   Full Copyright Statement . . . . . . . . . . . . . . . . . . . . . 13

Duric & Andersen              Experimental                      [Page 1]
RFC 3952           RTP Payload Format for iLBC Speech      December 2004

1.  Introduction

   This document describes how compressed iLBC speech, as produced by
   the iLBC codec [1], may be formatted for use as an RTP payload type.
   Methods are provided to packetize the codec data frames into RTP
   packets.  The sender may send one or more codec data frames per
   packet depending on the application scenario or based on the
   transport network condition, bandwidth restriction, delay
   requirements and packet-loss tolerance.

   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 BCP 14, RFC 2119 [2].

2.  Background

   Global IP Sound (GIPS) has developed a speech compression algorithm
   for use in IP based communications [1].  The iLBC codec enables
   graceful speech quality degradation in the case of lost frames, which
   occurs in connection with lost or delayed IP packets.

   This codec is suitable for real time communications such as,
   telephony and videoconferencing, streaming audio, archival and
   messaging.

   The iLBC codec [1] is an algorithm that compresses each basic frame
   (20 ms or 30 ms) of 8000 Hz, 16-bit sampled input speech, into output
   frames with rate of 400 bits for 30 ms basic frame size and 304 bits
   for 20 ms basic frame size.

   The codec supports two basic frame lengths: 30 ms at 13.33 kbit/s and
   20 ms at 15.2 kbit/s, using a block independent linear-predictive
   coding (LPC) algorithm.  When the codec operates at block lengths of
   20 ms, it produces 304 bits per block which MUST be packetized in 38
   bytes.  Similarly, for block lengths of 30 ms it produces 400 bits
   per block which MUST be packetized in 50 bytes.  This algorithm
   results in a speech coding system with a controlled response to
   packet losses similar to what is known from pulse code modulation
   (PCM) with a packet loss concealment (PLC), such as ITU-T G711
   standard [7], which operates at a fixed bit rate of 64 kbit/s.  At
   the same time, this algorithm enables fixed bit rate coding with a
   quality-versus-bit rate tradeoff close to what is known from code-
   excited linear prediction (CELP).
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