📄 rfc3267.txt
字号:
Network Working Group J. Sjoberg
Request for Comments: 3267 M. Westerlund
Category: Standards Track Ericsson
A. Lakaniemi
Nokia
Q. Xie
Motorola
June 2002
Real-Time Transport Protocol (RTP) Payload Format and File Storage
Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate
Wideband (AMR-WB) Audio Codecs
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 (2002). All Rights Reserved.
Abstract
This document specifies a real-time transport protocol (RTP) payload
format to be used for Adaptive Multi-Rate (AMR) and Adaptive Multi-
Rate Wideband (AMR-WB) encoded speech signals. The payload format is
designed to be able to interoperate with existing AMR and AMR-WB
transport formats on non-IP networks. In addition, a file format is
specified for transport of AMR and AMR-WB speech data in storage mode
applications such as email. Two separate MIME type registrations are
included, one for AMR and one for AMR-WB, specifying use of both the
RTP payload format and the storage format.
Sjoberg, et. al. Standards Track [Page 1]
RFC 3267 RTP Payload Format for AMR and AMR-WB June 2002
Table of Contents
1. Introduction.................................................... 3
2. Conventions and Acronyms........................................ 3
3. Background on AMR/AMR-WB and Design Principles.................. 4
3.1. The Adaptive Multi-Rate (AMR) Speech Codec.................. 4
3.2. The Adaptive Multi-Rate Wideband (AMR-WB) Speech Codec...... 5
3.3. Multi-rate Encoding and Mode Adaptation..................... 5
3.4. Voice Activity Detection and Discontinuous Transmission..... 6
3.5. Support for Multi-Channel Session........................... 6
3.6. Unequal Bit-error Detection and Protection.................. 7
3.6.1. Applying UEP and UED in an IP Network................... 7
3.7. Robustness against Packet Loss.............................. 9
3.7.1. Use of Forward Error Correction (FEC)................... 9
3.7.2. Use of Frame Interleaving...............................11
3.8. Bandwidth Efficient or Octet-aligned Mode...................11
3.9. AMR or AMR-WB Speech over IP scenarios......................12
4. AMR and AMR-WB RTP Payload Formats..............................14
4.1. RTP Header Usage............................................14
4.2. Payload Structure...........................................16
4.3. Bandwidth-Efficient Mode....................................16
4.3.1. The Payload Header......................................16
4.3.2. The Payload Table of Contents...........................17
4.3.3. Speech Data.............................................19
4.3.4. Algorithm for Forming the Payload.......................20
4.3.5 Payload Examples.........................................21
4.3.5.1. Single Channel Payload Carrying a Single Frame...21
4.3.5.2. Single Channel Payload Carrying Multiple Frames..22
4.3.5.3. Multi-Channel Payload Carrying Multiple Frames...23
4.4. Octet-aligned Mode..........................................25
4.4.1. The Payload Header......................................25
4.4.2. The Payload Table of Contents and Frame CRCs............26
4.4.2.1. Use of Frame CRC for UED over IP....................28
4.4.3. Speech Data.............................................30
4.4.4. Methods for Forming the Payload.........................30
4.4.5. Payload Examples........................................32
4.4.5.1. Basic Single Channel Payload Carrying
Multiple Frames..................................32
4.4.5.2. Two Channel Payload with CRC, Interleaving,
and Robust-sorting...............................32
4.5. Implementation Considerations...............................33
5. AMR and AMR-WB Storage Format...................................34
5.1. Single Channel Header.......................................34
5.2. Multi-channel Header........................................35
5.3. Speech Frames...............................................36
6. Congestion Control..............................................37
7. Security Considerations.........................................37
7.1. Confidentiality.............................................37
Sjoberg, et. al. Standards Track [Page 2]
RFC 3267 RTP Payload Format for AMR and AMR-WB June 2002
7.2. Authentication..............................................38
7.3. Decoding Validation.........................................38
8. Payload Format Parameters.......................................38
8.1. AMR MIME Registration.......................................39
8.2. AMR-WB MIME Registration....................................41
8.3. Mapping MIME Parameters into SDP............................44
9. IANA Considerations.............................................45
10. Acknowledgements...............................................45
11. References.....................................................45
11.1 Informative References......................................46
12. Authors' Addresses.............................................48
13. Full Copyright Statement.......................................49
1. Introduction
This document specifies the payload format for packetization of AMR
and AMR-WB encoded speech signals into the Real-time Transport
Protocol (RTP) [8]. The payload format supports transmission of
multiple channels, multiple frames per payload, the use of fast codec
mode adaptation, robustness against packet loss and bit errors, and
interoperation with existing AMR and AMR-WB transport formats on
non-IP networks, as described in Section 3.
The payload format itself is specified in Section 4. A related file
format is specified in Section 5 for transport of AMR and AMR-WB
speech data in storage mode applications such as email. In Section
8, two separate MIME type registrations are provided, one for AMR and
one for AMR-WB.
Even though this RTP payload format definition supports the transport
of both AMR and AMR-WB speech, it is important to remember that AMR
and AMR-WB are two different codecs and they are always handled as
different payload types in RTP.
2. Conventions and Acronyms
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 [5].
The following acronyms are used in this document:
3GPP - the Third Generation Partnership Project
AMR - Adaptive Multi-Rate Codec
AMR-WB - Adaptive Multi-Rate Wideband Codec
CMR - Codec Mode Request
CN - Comfort Noise
DTX - Discontinuous Transmission
Sjoberg, et. al. Standards Track [Page 3]
RFC 3267 RTP Payload Format for AMR and AMR-WB June 2002
ETSI - European Telecommunications Standards Institute
FEC - Forward Error Correction
SCR - Source Controlled Rate Operation
SID - Silence Indicator (the frames containing only CN
parameters)
VAD - Voice Activity Detection
UED - Unequal Error Detection
UEP - Unequal Error Protection
The term "frame-block" is used in this document to describe the
time-synchronized set of speech frames in a multi-channel AMR or
AMR-WB session. In particular, in an N-channel session, a frame-
block will contain N speech frames, one from each of the channels,
and all N speech frames represents exactly the same time period.
3. Background on AMR/AMR-WB and Design Principles
AMR and AMR-WB were originally designed for circuit-switched mobile
radio systems. Due to their flexibility and robustness, they are
also suitable for other real-time speech communication services over
packet-switched networks such as the Internet.
Because of the flexibility of these codecs, the behavior in a
particular application is controlled by several parameters that
select options or specify the acceptable values for a variable.
These options and variables are described in general terms at
appropriate points in the text of this specification as parameters to
be established through out-of-band means. In Section 8, all of the
parameters are specified in the form of MIME subtype registrations
for the AMR and AMR-WB encodings. The method used to signal these
parameters at session setup or to arrange prior agreement of the
participants is beyond the scope of this document; however, Section
8.3 provides a mapping of the parameters into the Session Description
Protocol (SDP) [11] for those applications that use SDP.
3.1. The Adaptive Multi-Rate (AMR) Speech Codec
The AMR codecs was originally developed and standardized by the
European Telecommunications Standards Institute (ETSI) for GSM
cellular systems. It is now chosen by the Third Generation
Partnership Project (3GPP) as the mandatory codec for third
generation (3G) cellular systems [1].
The AMR codec is a multi-mode codec that supports 8 narrow band
speech encoding modes with bit rates between 4.75 and 12.2 kbps. The
sampling frequency used in AMR is 8000 Hz and the speech encoding is
performed on 20 ms speech frames. Therefore, each encoded AMR speech
frame represents 160 samples of the original speech.
Sjoberg, et. al. Standards Track [Page 4]
RFC 3267 RTP Payload Format for AMR and AMR-WB June 2002
Among the 8 AMR encoding modes, three are already separately adopted
as standards of their own. Particularly, the 6.7 kbps mode is
adopted as PDC-EFR [14], the 7.4 kbps mode as IS-641 codec in TDMA
[13], and the 12.2 kbps mode as GSM-EFR [12].
3.2. The Adaptive Multi-Rate Wideband (AMR-WB) Speech Codec
The Adaptive Multi-Rate Wideband (AMR-WB) speech codec [3] was
originally developed by 3GPP to be used in GSM and 3G cellular
systems.
Similar to AMR, the AMR-WB codec is also a multi-mode speech codec.
AMR-WB supports 9 wide band speech coding modes with respective bit
rates ranging from 6.6 to 23.85 kbps. The sampling frequency used in
AMR-WB is 16000 Hz and the speech processing is performed on 20 ms
frames. This means that each AMR-WB encoded frame represents 320
speech samples.
3.3. Multi-rate Encoding and Mode Adaptation
The multi-rate encoding (i.e., multi-mode) capability of AMR and
AMR-WB is designed for preserving high speech quality under a wide
range of transmission conditions.
With AMR or AMR-WB, mobile radio systems are able to use available
bandwidth as effectively as possible. E.g., in GSM it is possible to
dynamically adjust the speech encoding rate during a session so as to
continuously adapt to the varying transmission conditions by dividing
the fixed overall bandwidth between speech data and error protective
coding to enable best possible trade-off between speech compression
rate and error tolerance. To perform mode adaptation, the decoder
(speech receiver) needs to signal the encoder (speech sender) the new
mode it prefers. This mode change signal is called Codec Mode
Request or CMR.
Since in most sessions speech is sent in both directions between the
two ends, the mode requests from the decoder at one end to the
encoder at the other end are piggy-backed over the speech frames in
the reverse direction. In other words, there is no out-of-band
signaling needed for sending CMRs.
Every AMR or AMR-WB codec implementation is required to support all
the respective speech coding modes defined by the codec and must be
able to handle mode switching to any of the modes at any time.
However, some transport systems may impose limitations in the number
of modes supported and how often the mode can change due to bandwidth
Sjoberg, et. al. Standards Track [Page 5]
RFC 3267 RTP Payload Format for AMR and AMR-WB June 2002
limitations or other constraints. For this reason, the decoder is
allowed to indicate its acceptance of a particular mode or a subset
of the defined modes for the session using out-of-band means.
For example, the GSM radio link can only use a subset of at most four
different modes in a given session. This subset can be any
combination of the 8 AMR modes for an AMR session or any combination
of the 9 AMR-WB modes for an AMR-WB session.
Moreover, for better interoperability with GSM through a gateway, the
decoder is allowed to use out-of-band means to set the minimum number
of frames between two mode changes and to limit the mode change among
neighboring modes only.
Section 8 specifies a set of MIME parameters that may be used to
signal these mode adaptation controls at session setup.
3.4. Voice Activity Detection and Discontinuous Transmission
Both codecs support voice activity detection (VAD) and generation of
comfort noise (CN) parameters during silence periods. Hence, the
codecs have the option to reduce the number of transmitted bits and
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