RTP Payload Format for ITU-T Recommendation G.711.1
RFC 5391

Network Working Group                                         A. Sollaud
Request for Comments: 5391                                France Telecom
Category: Standards Track                                  November 2008

          RTP Payload Format for ITU-T Recommendation G.711.1

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) 2008 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents (http://trustee.ietf.org/
   license-info) in effect on the date of publication of this document.
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Abstract

   This document specifies a Real-time Transport Protocol (RTP) payload
   format to be used for the ITU Telecommunication Standardization
   Sector (ITU-T) G.711.1 audio codec.  Two media type registrations are
   also included.

Sollaud                     Standards Track                     [Page 1]
RFC 5391             RTP Payload Format for G.711.1        November 2008

Table of Contents

   1. Introduction ....................................................2
   2. Background ......................................................2
   3. RTP Header Usage ................................................3
   4. Payload Format ..................................................4
      4.1. Payload Header .............................................4
      4.2. Audio Data .................................................5
   5. Payload Format Parameters .......................................6
      5.1. PCMA-WB Media Type Registration ............................7
      5.2. PCMU-WB Media Type Registration ............................8
      5.3. Mapping to SDP Parameters ..................................9
           5.3.1. Offer-Answer Model Considerations ...................9
           5.3.2. Declarative SDP Considerations .....................11
   6. G.711 Interoperability .........................................11
   7. Congestion Control .............................................12
   8. Security Considerations ........................................12
   9. IANA Considerations ............................................12
   10. References ....................................................13
      10.1. Normative References .....................................13
      10.2. Informative References ...................................13

1.  Introduction

   The ITU Telecommunication Standardization Sector (ITU-T)
   Recommendation G.711.1 [ITU-G.711.1] is an embedded wideband
   extension of the Recommendation G.711 [ITU-G.711] audio codec.  This
   document specifies a payload format for packetization of G.711.1
   encoded audio signals into the Real-time Transport Protocol (RTP).

   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 [RFC2119].

2.  Background

   G.711.1 is a G.711 embedded wideband speech and audio coding
   algorithm operating at 64, 80, and 96 kbps.  At 64 kbps, G.711.1 is
   fully interoperable with G.711.  Hence, an efficient deployment in
   existing G.711-based Voice over IP (VoIP) infrastructures is
   foreseen.

   The codec operates on 5-ms frames, and the default sampling rate is
   16 kHz.  Input and output at 8 kHz are also supported for narrowband
   modes.

Sollaud                     Standards Track                     [Page 2]
RFC 5391             RTP Payload Format for G.711.1        November 2008

   The encoder produces an embedded bitstream structured in three layers
   corresponding to three available bit rates: 64, 80, and 96 kbps.  The
   bitstream can be truncated at the decoder side or by any component of
   the communication system to adjust, "on the fly", the bit rate to the
   desired value.

   The following table gives more details on these layers.

               +-------+------------------------+----------+
               | Layer | Description            | Bit rate |
               +-------+------------------------+----------+