





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.














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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



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     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



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      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.



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   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





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   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
   packets during silence periods to a minimum.  The operation of
   sending CN parameters at regular intervals during silence periods is
   usually called discontinuous transmission (DTX) or source controlled
   rate (SCR) operation.  The AMR or AMR-WB frames containing CN
   parameters are called Silence Indicator (SID) frames.  See more
   details about VAD and DTX functionality in [9] and [10].

3.5. Support for Multi-Channel Session

   Both the RTP payload format and the storage format defined in this
   document support multi-channel audio content (e.g., a stereophonic
   speech session).

   Although AMR and AMR-WB codecs themselves do not support encoding of
   multi-channel audio content into a single bit stream, they can be
   used to separately encode and decode each of the individual channels.

   To transport (or store) the separately encoded multi-channel content,
   the speech frames for all channels that are framed and encoded for
   the same 20 ms periods are logically collected in a frame-block.

   At the session setup, out-of-band signaling must be used to indicate
   the number of channels in the session and the order of the speech
   frames from different channels in each frame-block.  When using SDP
   for signaling, the number of channels is specified in the rtpmap




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   attribute and the order of channels carried in each frame-block is
   implied by the number of channels as specified in Section 4.1 in
   [24].

3.6. Unequal Bit-error Detection and Protection

   The speech bits encoded in each AMR or AMR-WB frame have different
   perceptual sensitivity to bit errors.  This property has been
   exploited in cellular systems to achieve better voice quality by
   using unequal error protection and detection (UEP and UED)
   mechanisms.

   The UEP/UED mechanisms focus the protection and detection of
   corrupted bits to the perceptually most sensitive bits in an AMR or
   AMR-WB frame.  In particular, speech bits in an AMR or AMR-WB frame
   are divided into class A, B, and C, where bits in class A are most
   sensitive and bits in class C least sensitive (see Table 1 below for
   AMR and [4] for AMR-WB).  A frame is only declared damaged if there
   are bit errors found in the most sensitive bits, i.e., the class A
   bits.  On the other hand, it is acceptable to have some bit errors in
   the other bits, i.e., class B and C bits.

                                    Class A   total speech
                  Index   Mode       bits       bits
                  ----------------------------------------
                    0     AMR 4.75   42         95
                    1     AMR 5.15   49        103
                    2     AMR 5.9    55        118
                    3     AMR 6.7    58        134
                    4     AMR 7.4    61        148
                    5     AMR 7.95   75        159
                    6     AMR 10.2   65        204
                    7     AMR 12.2   81        244
                    8     AMR SID    39         39

          Table 1.  The number of class A bits for the AMR codec.

   Moreover, a damaged frame is still useful for error concealment at
   the decoder since some of the less sensitive bits can still be used.
   This approach can improve the speech quality compared to discarding
   the damaged frame.

3.6.1. Applying UEP and UED in an IP Network

   To take full advantage of the bit-error robustness of the AMR and
   AMR-WB codec, the RTP payload format is designed to facilitate
   UEP/UED in an IP network.  It should be noted however that the
   utilization of UEP and UED discussed below is OPTIONAL.



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   UEP/UED in an IP network can be achieved by detecting bit errors in
   class A bits and tolerating bit errors in class B/C bits of the AMR
   or AMR-WB frame(s) in each RTP payload.

   Today there exist some link layers that do not discard packets with
   bit errors, e.g., SLIP and some wireless links.  With the Internet
   traffic pattern shifting towards a more multimedia-centric one, more
   link layers of such nature may emerge in the future.  With transport
   layer support for partial checksums, for example those supported by
   UDP-Lite [15], bit error tolerant AMR and AMR-WB traffic could
   achieve better performance over these types of links.

   There are at least two basic approaches for carrying AMR and AMR-WB
   traffic over bit error tolerant IP networks:

   1) Utilizing a partial checksum to cover headers and the most
      important speech bits of the payload.  It is recommended that at
      least all class A bits are covered by the checksum.

   2) Utilizing a partial checksum to only cover headers, but a frame
      CRC to cover the class A bits of each speech frame in the RTP
      payload.

   In either approach, at least part of the class B/C bits are left
   without error-check and thus bit error tolerance is achieved.

      Note, it is still important that the network designer pay
      attention to the class B and C residual bit error rate.  Though
      less sensitive to errors than class A bits, class B and C bits are
      not insignificant and undetected errors in these bits cause
      degradation in speech quality.  An example of residual error rates
      considered acceptable for AMR in UMTS can be found in [20] and for
      AMR-WB in [21].

   The application interface to the UEP/UED transport protocol (e.g.,
   UDP-Lite) may not provide any control over the link error rate,
   especially in a gateway scenario.  Therefore, it is incumbent upon
   the designer of a node with a link interface of this type to choose a
   residual bit error rate that is low enough to support applications
   such as AMR encoding when transmitting packets of a UEP/UED transport
   protocol.

   Approach 1 is a bit efficient, flexible and simple way, but comes
   with two disadvantages, namely, a) bit errors in protected speech
   bits will cause the payload to be discarded, and b) when transporting
   multiple frames in a payload there is the possibility that a single
   bit error in protected bits will cause all the frames to be
   discarded.



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   These disadvantages can be avoided, if needed, with some overhead in
   the form of a frame-wise CRC (Approach 2).  In problem a), the CRC
   makes it possible to detect bit errors in class A bits and use the
   frame for error concealment, which gives a small improvement in
   speech quality.  For b), when transporting multiple frames in a
   payload, the CRCs remove the possibility that a single bit error in a
   class A bit will cause all the frames to be discarded.  Avoiding that
   gives an improvement in speech quality when transporting multiple
   frames over links subject to bit errors.

   The choice between the above two approaches must be made based on the
   available bandwidth, and desired tolerance to bit errors.  Neither
   solution is appropriate to all cases.  Section 8 defines parameters
   that may be used at session setup to select between these approaches.

3.7. Robustness against Packet Loss

   The payload format supports several means, including forward error
   correction (FEC) and frame interleaving, to increase robustness
   against packet loss.

3.7.1. Use of Forward Error Correction (FEC)

   The simple scheme of repetition of previously sent data is one way of
   achieving FEC.  Another possible scheme which is more bandwidth
   efficient is to use payload external FEC, e.g., RFC2733 [19], which
   generates extra packets containing repair data.  The whole payload
   can also be sorted in sensitivity order to support external FEC
   schemes using UEP.  There is also a work in progress on a generic
   version of such a scheme [18] that can be applied to AMR or AMR-WB
   payload transport.

   With AMR or AMR-WB, it is possible to use the multi-rate capability
   of the codec to send redundant copies of the same mode or of another
   mode, e.g., one with lower-bandwidth.  We describe such a scheme
   next.















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   This involves the simple retransmission of previously transmitted
   frame-blocks together with the current frame-block(s).  This is done
   by using a sliding window to group the speech frame-blocks to send in
   each payload.  Figure 1 below shows us an example.

   --+--------+--------+--------+--------+--------+--------+--------+--
     | f(n-2) | f(n-1) |  f(n)  | f(n+1) | f(n+2) | f(n+3) | f(n+4) |
   --+--------+--------+--------+--------+--------+--------+--------+--

     <---- p(n-1) ---->
              <----- p(n) ----->
                       <---- p(n+1) ---->
                                <---- p(n+2) ---->
                                         <---- p(n+3) ---->
                                                  <---- p(n+4) ---->

              Figure 1: An example of redundant transmission.

   In this example each frame-block is retransmitted one time in the
   following RTP payload packet.  Here, f(n-2)..f(n+4) denotes a
   sequence of speech frame-blocks and p(n-1)..p(n+4) a sequence of
   payload packets.

   The use of this approach does not require signaling at the session
   setup.  In other words, the speech sender can choose to use this
   scheme without consulting the receiver.  This is because a packet
   containing redundant frames will not look different from a packet
   with only new frames.  The receiver may receive multiple copies or
   versions (encoded with different modes) of a frame for a certain
   timestamp if no packet is lost.  If multiple versions of the same
   speech frame are received, it is recommended that the mode with the
   highest rate be used by the speech decoder.

   This redundancy scheme provides the same functionality as the one
   described in RFC 2198 "RTP Payload for Redundant Audio Data" [24].
   In most cases the mechanism in this payload format is more efficient
   and simpler than requiring both endpoints to support RFC 2198 in
   addition.  There are two situations in which use of RFC 2198 is
   indicated: if the spread in time required between the primary and
   redundant encodings is larger than 5 frame times, the bandwidth
   overhead of RFC 2198 will be lower; or, if a non-AMR codec is desired
   for the redundant encoding, the AMR payload format won't be able to
   carry it.

   The sender is responsible for selecting an appropriate amount of
   redundancy based on feedback about the channel, e.g., in RTCP
   receiver reports.  A sender should not base selection of FEC on the
   CMR, as this parameter most probably was set based on none-IP



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   information, e.g., radio link performance measures.  The sender is
   also responsible for avoiding congestion, which may be exacerbated by
   redundancy (see Section 6 for more details).

3.7.2. Use of Frame Interleaving

   To decrease protocol overhead, the payload design allows several
   speech frame-blocks be encapsulated into a single RTP packet.  One of
   the drawbacks of such an approach is that in case of packet loss this
   means loss of several consecutive speech frame-blocks, which usually
   causes clearly audible distortion in the reconstructed speech.
   Interleaving of frame-blocks can improve the speech quality in such
   cases by distributing the consecutive losses into a series of single
   frame-block losses.  However, interleaving and bundling several
   frame-blocks per payload will also increase end-to-end delay and is
   therefore not appropriate for all types of applications.  Streaming
   applications will most likely be able to exploit interleaving to
   improve speech quality in lossy transmission conditions.

   This payload design supports the use of frame interleaving as an
   option.  For the encoder (speech sender) to use frame interleaving in
   its outbound RTP packets for a given session, the decoder (speech
   receiver) needs to indicate its support via out-of-band means (see
   Section 8).

3.8. Bandwidth Efficient or Octet-aligned Mode

   For a given session, the payload format can be either bandwidth
   efficient or octet aligned, depending on the mode of operation that
   is established for the session via out-of-band means.

   In the octet-aligned format, all the fields in a payload, including
   payload header, table of contents entries, and speech frames
   themselves, are individually aligned to octet boundaries to make
   implementations efficient.  In the bandwidth efficient format only
   the full payload is octet aligned, so fewer padding bits are added.

      Note, octet alignment of a field or payload means that the last
      octet is padded with zeroes in the least significant bits to fill
      the octet.  Also note that this padding is separate from padding
      indicated by the P bit in the RTP header.

   Between the two operation modes, only the octet-aligned mode has the
   capability to use the robust sorting, interleaving, and frame CRC to
   make the speech transport robust to packet loss and bit errors.






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3.9. AMR or AMR-WB Speech over IP scenarios

   The primary scenario for this payload format is IP end-to-end between
   two terminals, as shown in Figure 2.  This payload format is expected
   to be useful for both conversational and streaming services.

                +----------+                         +----------+
                |          |    IP/UDP/RTP/AMR or    |          |
                | TERMINAL |<----------------------->| TERMINAL |
                |          |    IP/UDP/RTP/AMR-WB    |          |
                +----------+                         +----------+

                   Figure 2: IP terminal to IP terminal scenario

   A conversational service puts requirements on the payload format.
   Low delay is one very important factor, i.e., few speech frame-blocks
   per payload packet.  Low overhead is also required when the payload
   format traverses low bandwidth links, especially as the frequency of
   packets will be high.  For low bandwidth links it also an advantage
   to support UED which allows a link provider to reduce delay and
   packet loss or to reduce the utilization of link resources.

   Streaming service has less strict real-time requirements and
   therefore can use a larger number of frame-blocks per packet than
   conversational service.  This reduces the overhead from IP, UDP, and
   RTP headers.  However, including several frame-blocks per packet
   makes the transmission more vulnerable to packet loss, so
   interleaving may be used to reduce the effect packet loss will have
   on speech quality.  A streaming server handling a large number of
   clients also needs a payload format that requires as few resources as
   possible when doing packetization.  The octet-aligned and
   interleaving modes require the least amount of resources, while CRC,
   robust sorting, and bandwidth efficient modes have higher demands.

   Another scenario occurs when AMR or AMR-WB encoded speech will be
   transmitted from a non-IP system (e.g., a GSM or 3GPP network) to an
   IP/UDP/RTP VoIP terminal, and/or vice versa, as depicted in Figure 3.














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          AMR or AMR-WB
          over
          I.366.{2,3} or +------+                        +----------+
          3G Iu or       |      |   IP/UDP/RTP/AMR or    |          |
          <------------->|  GW  |<---------------------->| TERMINAL |
          GSM Abis       |      |   IP/UDP/RTP/AMR-WB    |          |
          etc.           +------+                        +----------+
                             |
           GSM/3GPP network  |           IP network
                             |

                     Figure 3: GW to VoIP terminal scenario

   In such a case, it is likely that the AMR or AMR-WB frame is
   packetized in a different way in the non-IP network and will need to
   be re-packetized into RTP at the gateway.  Also, speech frames from
   the non-IP network may come with some UEP/UED information (e.g., a
   frame quality indicator) that will need to be preserved and forwarded
   on to the decoder along with the speech bits.  This is specified in
   Section 4.3.2.

   AMR's capability to do fast mode switching is exploited in some non-
   IP networks to optimize speech quality.  To preserve this
   functionality in scenarios including a gateway to an IP network, a
   codec mode request (CMR) field is needed.  The gateway will be
   responsible for forwarding the CMR between the non-IP and IP parts in
   both directions.  The IP terminal should follow the CMR forwarded by
   the gateway to optimize speech quality going to the non-IP decoder.
   The mode control algorithm in the gateway must accommodate the delay
   imposed by the IP network on the response to CMR by the IP terminal.

   The IP terminal should not set the CMR (see Section 4.3.1), but the
   gateway can set the CMR value on frames going toward the encoder in
   the non-IP part to optimize speech quality from that encoder to the
   gateway.  The gateway can alternatively set a lower CMR value, if
   desired, as one means to control congestion on the IP network.















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   A third likely scenario is that IP/UDP/RTP is used as transport
   between two non-IP systems, i.e., IP is originated and terminated in
   gateways on both sides of the IP transport, as illustrated in Figure
   4 below.

   AMR or AMR-WB                                        AMR or AMR-WB
   over                                                 over
   I.366.{2,3} or +------+                     +------+ I.366.{2,3} or
   3G Iu or       |      |  IP/UDP/RTP/AMR or  |      | 3G Iu or
   <------------->|  GW  |<------------------->|  GW  |<------------->
   GSM Abis       |      |  IP/UDP/RTP/AMR-WB  |      | GSM Abis
   etc.           +------+                     +------+ etc.
                      |                           |
    GSM/3GPP network  |          IP network       |  GSM/3GPP network
                      |                           |

                        Figure 4: GW to GW scenario

   This scenario requires the same mechanisms for preserving UED/UEP and
   CMR information as in the single gateway scenario.  In addition, the
   CMR value may be set in packets received by the gateways on the IP
   network side.  The gateway should forward to the non-IP side a CMR
   value that is the minimum of three values:

      -  the CMR value it receives on the IP side;

      -  the CMR value it calculates based on its reception quality on
         the non-IP side; and

      - a CMR value it may choose for congestion control of transmission
         on the IP side.

   The details of the control algorithm are left to the implementation.

4. AMR and AMR-WB RTP Payload Formats

   The AMR and AMR-WB payload formats have identical structure, so they
   are specified together.  The only differences are in the types of
   codec frames contained in the payload.  The payload format consists
   of the RTP header, payload header and payload data.

4.1. RTP Header Usage

   The format of the RTP header is specified in [8].  This payload
   format uses the fields of the header in a manner consistent with that
   specification.





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   The RTP timestamp corresponds to the sampling instant of the first
   sample encoded for the first frame-block in the packet.  The
   timestamp clock frequency is the same as the sampling frequency, so
   the timestamp unit is in samples.

   The duration of one speech frame-block is 20 ms for both AMR and
   AMR-WB.  For AMR, the sampling frequency is 8 kHz, corresponding to
   160 encoded speech samples per frame from each channel.  For AMR-WB,
   the sampling frequency is 16 kHz, corresponding to 320 samples per
   frame from each channel.  Thus, the timestamp is increased by 160 for
   AMR and 320 for AMR-WB for each consecutive frame-block.

   A packet may contain multiple frame-blocks of encoded speech or
   comfort noise parameters.  If interleaving is employed, the frame-
   blocks encapsulated into a payload are picked according to the
   interleaving rules as defined in Section 4.4.1.  Otherwise, each
   packet covers a period of one or more contiguous 20 ms frame-block
   intervals.  In case the data from all the channels for a particular
   frame-block in the period is missing, for example at a gateway from
   some other transport format, it is possible to indicate that no data
   is present for that frame-block rather than breaking a multi-frame-
   block packet into two, as explained in Section 4.3.2.

   To allow for error resiliency through redundant transmission, the
   periods covered by multiple packets MAY overlap in time.  A receiver
   MUST be prepared to receive any speech frame multiple times, either
   in exact duplicates, or in different AMR rate modes, or with data
   present in one packet and not present in another.  If multiple
   versions of the same speech frame are received, it is RECOMMENDED
   that the mode with the highest rate be used by the speech decoder.  A
   given frame MUST NOT be encoded as speech in one packet and comfort
   noise parameters in another.

   The payload is always made an integral number of octets long by
   padding with zero bits if necessary.  If additional padding is
   required to bring the payload length to a larger multiple of octets
   or for some other purpose, then the P bit in the RTP in the header
   may be set and padding appended as specified in [8].

   The RTP header marker bit (M) SHALL be set to 1 if the first frame-
   block carried in the packet contains a speech frame which is the
   first in a talkspurt.  For all other packets the marker bit SHALL be
   set to zero (M=0).








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   The assignment of an RTP payload type for this new packet format is
   outside the scope of this document, and will not be specified here.
   It is expected that the RTP profile under which this payload format
   is being used will assign a payload type for this encoding or specify
   that the payload type is to be bound dynamically.

4.2. Payload Structure

   The complete payload consists of a payload header, a payload table of
   contents, and speech data representing one or more speech frame-
   blocks.  The following diagram shows the general payload format
   layout:

   +----------------+-------------------+----------------
   | payload header | table of contents | speech data ...
   +----------------+-------------------+----------------

   Payloads containing more than one speech frame-block are called
   compound payloads.

   The following sections describe the variations taken by the payload
   format depending on whether the AMR session is set up to use the
   bandwidth-efficient mode or octet-aligned mode and any of the
   OPTIONAL functions for robust sorting, interleaving, and frame CRCs.
   Implementations SHOULD support both bandwidth-efficient and octet-
   aligned operation to increase interoperability.

4.3. Bandwidth-Efficient Mode

4.3.1. The Payload Header

   In bandwidth-efficient mode, the payload header simply consists of a
   4 bit codec mode request:

    0 1 2 3
   +-+-+-+-+
   |  CMR  |
   +-+-+-+-+

   CMR (4 bits): Indicates a codec mode request sent to the speech
      encoder at the site of the receiver of this payload.  The value of
      the CMR field is set to the frame type index of the corresponding
      speech mode being requested.  The frame type index may be 0-7 for
      AMR, as defined in Table 1a in [2], or 0-8 for AMR-WB, as defined
      in Table 1a in [4].  CMR value 15 indicates that no mode request
      is present, and other values are for future use.





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   The mode request received in the CMR field is valid until the next
   CMR is received, i.e., a newly received CMR value overrides the
   previous one.  Therefore, if a terminal continuously wishes to
   receive frames in the same mode X, it needs to set CMR=X for all its
   outbound payloads, and if a terminal has no preference in which mode
   to receive, it SHOULD set CMR=15 in all its outbound payloads.

   If receiving a payload with a CMR value which is not a speech mode or
   NO_DATA, the CMR MUST be ignored by the receiver.

   In a multi-channel session, CMR SHOULD be interpreted by the receiver
   of the payload as the desired encoding mode for all the channels in
   the session.

   An IP end-point SHOULD NOT set the CMR based on packet losses or
   other congestion indications, for several reasons:

      -  The other end of the IP path may be a gateway to a non-IP
         network (such as a radio link) that needs to set the CMR field
         to optimize performance on that network.

      -  Congestion on the IP network is managed by the IP sender, in
         this case at the other end of the IP path.  Feedback about
         congestion SHOULD be provided to that IP sender through RTCP or
         other means, and then the sender can choose to avoid congestion
         using the most appropriate mechanism.  That may include
         adjusting the codec mode, but also includes adjusting the level
         of redundancy or number of frames per packet.

   The encoder SHOULD follow a received mode request, but MAY change to
   a lower-numbered mode if it so chooses, for example to control
   congestion.

   The CMR field MUST be set to 15 for packets sent to a multicast
   group.  The encoder in the speech sender SHOULD ignore mode requests
   when sending speech to a multicast session but MAY use RTCP feedback
   information as a hint that a mode change is needed.

   The codec mode selection MAY be restricted by a session parameter to
   a subset of the available modes.  If so, the requested mode MUST be
   among the signalled subset (see Section 8).

4.3.2. The Payload Table of Contents

   The table of contents (ToC) consists of a list of ToC entries, each
   representing a speech frame.





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   In bandwidth-efficient mode, a ToC entry takes the following format:

    0 1 2 3 4 5
   +-+-+-+-+-+-+
   |F|  FT   |Q|
   +-+-+-+-+-+-+

   F (1 bit): If set to 1, indicates that this frame is followed by
      another speech frame in this payload; if set to 0, indicates that
      this frame is the last frame in this payload.

   FT (4 bits): Frame type index, indicating either the AMR or AMR-WB
      speech coding mode or comfort noise (SID) mode of the
      corresponding frame carried in this payload.

   The value of FT is defined in Table 1a in [2] for AMR and in Table 1a
   in [4] for AMR-WB.  FT=14 (SPEECH_LOST, only available for AMR-WB)
   and FT=15 (NO_DATA) are used to indicate frames that are either lost
   or not being transmitted in this payload, respectively.

   NO_DATA (FT=15) frame could mean either that there is no data
   produced by the speech encoder for that frame or that no data for
   that frame is transmitted in the current payload (i.e., valid data
   for that frame could be sent in either an earlier or later packet).

   If receiving a ToC entry with a FT value in the range 9-14 for AMR or
   10-13 for AMR-WB the whole packet SHOULD be discarded.  This is to
   avoid the loss of data synchronization in the depacketization
   process, which can result in a huge degradation in speech quality.

   Note that packets containing only NO_DATA frames SHOULD NOT be
   transmitted.  Also, frame-blocks containing only NO_DATA frames at
   the end of a packet SHOULD NOT be transmitted, except in the case of
   interleaving.  The AMR SCR/DTX is described in [6] and AMR-WB SCR/DTX
   in [7].

   The extra comfort noise frame types specified in table 1a in [2]
   (i.e., GSM-EFR CN, IS-641 CN, and PDC-EFR CN) MUST NOT be used in
   this payload format because the standardized AMR codec is only
   required to implement the general AMR SID frame type and not those
   that are native to the incorporated encodings.

   Q (1 bit): Frame quality indicator.  If set to 0, indicates the
      corresponding frame is severely damaged and the receiver should
      set the RX_TYPE (see [6]) to either SPEECH_BAD or SID_BAD
      depending on the frame type (FT).





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   The frame quality indicator is included for interoperability with the
   ATM payload format described in ITU-T I.366.2, the UMTS Iu interface
   [16], as well as other transport formats.  The frame quality
   indicator enables damaged frames to be forwarded to the speech
   decoder for error concealment.  This can improve the speech quality
   comparing to dropping the damaged frames.  See Section 4.4.2.1 for
   more details.

   For multi-channel sessions, the ToC entries of all frames from a
   frame-block are placed in the ToC in consecutive order as defined in
   Section 4.1 in [24].  When multiple frame-blocks are present in a
   packet in bandwidth-efficient mode, they will be placed in the packet
   in order of their creation time.

   Therefore, with N channels and K speech frame-blocks in a packet,
   there MUST be N*K entries in the ToC, and the first N entries will be
   from the first frame-block, the second N entries will be from the
   second frame-block, and so on.

   The following figure shows an example of a ToC of three entries in a
   single channel session using bandwidth efficient mode.

    0                   1
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |1|  FT   |Q|1|  FT   |Q|0|  FT   |Q|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   Below is an example of how the ToC entries will appear in the ToC of
   a packet carrying 3 consecutive frame-blocks in a session with two
   channels (L and R).

   +----+----+----+----+----+----+
   | 1L | 1R | 2L | 2R | 3L | 3R |
   +----+----+----+----+----+----+
   |<------->|<------->|<------->|
     Frame-    Frame-    Frame-
     Block 1   Block 2   Block 3

4.3.3. Speech Data

   Speech data of a payload contains one or more speech frames or
   comfort noise frames, as described in the ToC of the payload.

      Note, for ToC entries with FT=14 or 15, there will be no
      corresponding speech frame present in the speech data.





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   Each speech frame represents 20 ms of speech encoded with the mode
   indicated in the FT field of the corresponding ToC entry.  The length
   of the speech frame is implicitly defined by the mode indicated in
   the FT field.  The order and numbering notation of the bits are as
   specified for Interface Format 1 (IF1) in [2] for AMR and [4] for
   AMR-WB.  As specified there, the bits of speech frames have been
   rearranged in order of decreasing sensitivity, while the bits of
   comfort noise frames are in the order produced by the encoder.  The
   resulting bit sequence for a frame of length K bits is denoted d(0),
   d(1), ..., d(K-1).

4.3.4. Algorithm for Forming the Payload

   The complete RTP payload in bandwidth-efficient mode is formed by
   packing bits from the payload header, table of contents, and speech
   frames, in order as defined by their corresponding ToC entries in the
   ToC list, contiguously into octets beginning with the most
   significant bits of the fields and the octets.

   To be precise, the four-bit payload header is packed into the first
   octet of the payload with bit 0 of the payload header in the most
   significant bit of the octet.  The four most significant bits
   (numbered 0-3) of the first ToC entry are packed into the least
   significant bits of the octet, ending with bit 3 in the least
   significant bit.  Packing continues in the second octet with bit 4 of
   the first ToC entry in the most significant bit of the octet.  If
   more than one frame is contained in the payload, then packing
   continues with the second and successive ToC entries.  Bit 0 of the
   first data frame follows immediately after the last ToC bit,
   proceeding through all the bits of the frame in numerical order.
   Bits from any successive frames follow contiguously in numerical
   order for each frame and in consecutive order of the frames.

   If speech data is missing for one or more speech frame within the
   sequence, because of, for example, DTX, a ToC entry with FT set to
   NO_DATA SHALL be included in the ToC for each of the missing frames,
   but no data bits are included in the payload for the missing frame
   (see Section 4.3.5.2 for an example).













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4.3.5 Payload Examples

4.3.5.1. Single Channel Payload Carrying a Single Frame

   The following diagram shows a bandwidth-efficient AMR payload from a
   single channel session carrying a single speech frame-block.

   In the payload, no specific mode is requested (CMR=15), the speech
   frame is not damaged at the IP origin (Q=1), and the coding mode is
   AMR 7.4 kbps (FT=4).  The encoded speech bits, d(0) to d(147), are
   arranged in descending sensitivity order according to [2].  Finally,
   two zero bits are added to the end as padding to make the payload
   octet aligned.

    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | CMR=15|0| FT=4  |1|d(0)                                       |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                     d(147)|P|P|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
























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4.3.5.2. Single Channel Payload Carrying Multiple Frames

   The following diagram shows a single channel, bandwidth efficient
   compound AMR-WB payload that contains four frames, of which one has
   no speech data.  The first frame is a speech frame at 6.6 kbps mode
   (FT=0) that is composed of speech bits d(0) to d(131).  The second
   frame is an AMR-WB SID frame (FT=9), consisting of bits g(0) to
   g(39).  The third frame is NO_DATA frame and does not carry any
   speech information, it is represented in the payload by its ToC
   entry.  The fourth frame in the payload is a speech frame at 8.85
   kpbs mode (FT=1), it consists of speech bits h(0) to h(176).

   As shown below, the payload carries a mode request for the encoder on
   the receiver's side to change its future coding mode to AMR-WB 8.85
   kbps (CMR=1).  None of the frames is damaged at IP origin (Q=1).  The
   encoded speech and SID bits, d(0) to d(131), g(0) to g(39) and h(0)
   to h(176), are arranged in the payload in descending sensitivity
   order according to [4]. (Note, no speech bits are present for the
   third frame).  Finally, seven 0s are padded to the end to make the
   payload octet aligned.

    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | CMR=1 |1| FT=0  |1|1| FT=9  |1|1| FT=15 |1|0| FT=1  |1|d(0)   |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                         d(131)|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |g(0)                                                           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          g(39)|h(0)                                           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                           h(176)|P|P|P|P|P|P|P|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+





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4.3.5.3. Multi-Channel Payload Carrying Multiple Frames

   The following diagram shows a two channel payload carrying 3 frame-
   blocks, i.e., the payload will contain 6 speech frames.

   In the payload all speech frames contain the same mode 7.4 kbit/s
   (FT=4) and are not damaged at IP origin.  The CMR is set to 15, i.e.,
   no specific mode is requested.  The two channels are defined as left
   (L) and right (R) in that order.  The encoded speech bits is
   designated dXY(0).. dXY(K-1), where X = block number, Y = channel,
   and K is the number of speech bits for that mode.  Exemplifying this,
   for frame-block 1 of the left channel the encoded bits are designated
   as d1L(0) to d1L(147).






































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    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | CMR=15|1|1L FT=4|1|1|1R FT=4|1|1|2L FT=4|1|1|2R FT=4|1|1|3L FT|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |4|1|0|3R FT=4|1|d1L(0)                                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                               d1L(147)|d1R(0) |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   : ...                                                           :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                       d1R(147)|d2L(0)                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   : ...                                                           :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |d2L(147|d2R(0)                                                 |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   : ...                                                           :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                       d2R(147)|d3L(0)         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   : ...                                                           :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |               d3L(147)|d3R(0)                                 |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   : ...                                                           :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                       d3R(147)|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
















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4.4. Octet-aligned Mode

4.4.1. The Payload Header

   In octet-aligned mode, the payload header consists of a 4 bit CMR, 4
   reserved bits, and optionally, an 8 bit interleaving header, as shown
   below:

    0                   1
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
   +-+-+-+-+-+-+-+-+- - - - - - - -
   |  CMR  |R|R|R|R|  ILL  |  ILP  |
   +-+-+-+-+-+-+-+-+- - - - - - - -

   CMR (4 bits): same as defined in section 4.3.1.

   R: is a reserved bit that MUST be set to zero.  All R bits MUST be
      ignored by the receiver.

   ILL (4 bits, unsigned integer): This is an OPTIONAL field that is
      present only if interleaving is signalled out-of-band for the
      session.  ILL=L indicates to the receiver that the interleaving
      length is L+1, in number of frame-blocks.

   ILP (4 bits, unsigned integer): This is an OPTIONAL field that is
      present only if interleaving is signalled.  ILP MUST take a value
      between 0 and ILL, inclusive, indicating the interleaving index
      for frame-blocks in this payload in the interleave group.  If the
      value of ILP is found greater than ILL, the payload SHOULD be
      discarded.

   ILL and ILP fields MUST be present in each packet in a session if
   interleaving is signalled for the session.  Interleaving MUST be
   performed on a frame-block basis (i.e., NOT on a frame basis) in a
   multi-channel session.

   The following example illustrates the arrangement of speech frame-
   blocks in an interleave group during an interleave session.  Here we
   assume ILL=L for the interleave group that starts at speech frame-
   block n.  We also assume that the first payload packet of the
   interleave group is s and the number of speech frame-blocks carried
   in each payload is N. Then we will have:

   Payload s (the first packet of this interleave group):
     ILL=L, ILP=0,
     Carry frame-blocks: n, n+(L+1), n+2*(L+1), ..., n+(N-1)*(L+1)

      Payload s+1 (the second packet of this interleave group):



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     ILL=L, ILP=1,
     frame-blocks: n+1, n+1+(L+1), n+1+2*(L+1), ..., n+1+(N-1)*(L+1)
       ...

   Payload s+L (the last packet of this interleave group):
     ILL=L, ILP=L,
     frame-blocks: n+L, n+L+(L+1), n+L+2*(L+1), ..., n+L+(N-1)*(L+1)

   The next interleave group will start at frame-block n+N*(L+1).

   There will be no interleaving effect unless the number of frame-
   blocks per packet (N) is at least 2.  Moreover, the number of frame-
   blocks per payload (N) and the value of ILL MUST NOT be changed
   inside an interleave group.  In other words, all payloads in an
   interleave group MUST have the same ILL and MUST contain the same
   number of speech frame-blocks.

   The sender of the payload MUST only apply interleaving if the
   receiver has signalled its use through out-of-band means.  Since
   interleaving will increase buffering requirements at the receiver,
   the receiver uses MIME parameter "interleaving=I" to set the maximum
   number of frame-blocks allowed in an interleaving group to I.

   When performing interleaving the sender MUST use a proper number of
   frame-blocks per payload (N) and ILL so that the resulting size of an
   interleave group is less or equal to I, i.e., N*(L+1)<=I.

4.4.2. The Payload Table of Contents and Frame CRCs

   The table of contents (ToC) in octet-aligned mode consists of a list
   of ToC entries where each entry corresponds to a speech frame carried
   in the payload and, optionally, a list of speech frame CRCs, i.e.,

   +---------------------+
   | list of ToC entries |
   +---------------------+
   | list of frame CRCs  | (optional)
    - - - - - - - - - - -

      Note, for ToC entries with FT=14 or 15, there will be no
      corresponding speech frame or frame CRC present in the payload.

   The list of ToC entries is organized in the same way as described for
   bandwidth-efficient mode in 4.3.2, with the following exception; when
   interleaving is used the frame-blocks in the ToC will almost never be
   placed consecutive in time.  Instead, the presence and order of the
   frame-blocks in a packet will follow the pattern described in 4.4.1.




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   The following example shows the ToC of three consecutive packets,
   each carrying 3 frame-blocks, in an interleaved two-channel session.
   Here, the two channels are left (L) and right (R) with L coming
   before R, and the interleaving length is 3 (i.e., ILL=2).  This makes
   the interleave group 9 frame-blocks large.

   Packet #1
   ---------

   ILL=2, ILP=0:
   +----+----+----+----+----+----+
   | 1L | 1R | 4L | 4R | 7L | 7R |
   +----+----+----+----+----+----+
   |<------->|<------->|<------->|
     Frame-    Frame-    Frame-
     Block 1   Block 4   Block 7

   Packet #2
   ---------

   ILL=2, ILP=1:
   +----+----+----+----+----+----+
   | 2L | 2R | 5L | 5R | 8L | 8R |
   +----+----+----+----+----+----+
   |<------->|<------->|<------->|
     Frame-    Frame-    Frame-
     Block 2   Block 5   Block 8

   Packet #3
   ---------

   ILL=2, ILP=2:
   +----+----+----+----+----+----+
   | 3L | 3R | 6L | 6R | 9L | 9R |
   +----+----+----+----+----+----+
   |<------->|<------->|<------->|
     Frame-    Frame-    Frame-
     Block 3   Block 6   Block 9

   A ToC entry takes the following format in octet-aligned mode:

    0 1 2 3 4 5 6 7
   +-+-+-+-+-+-+-+-+
   |F|  FT   |Q|P|P|
   +-+-+-+-+-+-+-+-+

   F (1 bit): see definition in Section 4.3.2.




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   FT (4 bits unsigned integer): see definition in Section 4.3.2.

   Q (1 bit): see definition in Section 4.3.2.

   P bits: padding bits, MUST be set to zero.

   The list of CRCs is OPTIONAL.  It only exists if the use of CRC is
   signalled out-of-band for the session.  When present, each CRC in the
   list is 8 bit long and corresponds to a speech frame (NOT a frame-
   block) carried in the payload.  Calculation and use of the CRC is
   specified in the next section.

4.4.2.1. Use of Frame CRC for UED over IP

   The general concept of UED/UEP over IP is discussed in Section 3.6.
   This section provides more details on how to use the frame CRC in the
   octet-aligned payload header together with a partial transport layer
   checksum to achieve UED.

   To achieve UED, one SHOULD use a transport layer checksum, for
   example, the one defined in UDP-Lite [15], to protect the RTP header,
   payload header, and table of contents bits in a payload.  The frame
   CRC, when used, MUST be calculated only over all class A bits in the
   frame.  Class B and C bits in the frame MUST NOT be included in the
   CRC calculation and SHOULD NOT be covered by the transport checksum.

      Note, the number of class A bits for various coding modes in AMR
      codec is specified as informative in [2] and is therefore copied
      into Table 1 in Section 3.6 to make it normative for this payload
      format.  The number of class A bits for various coding modes in
      AMR-WB codec is specified as normative in table 2 in [4], and the
      SID frame (FT=9) has 40 class A bits.  These definitions of class
      A bits MUST be used for this payload format.

   Packets SHOULD be discarded if the transport layer checksum detects
   errors.

   The receiver of the payload SHOULD examine the data integrity of the
   received class A bits by re-calculating the CRC over the received
   class A bits and comparing the result to the value found in the
   received payload header.  If the two values mismatch, the receiver
   SHALL consider the class A bits in the receiver frame damaged and
   MUST clear the Q flag of the frame (i.e., set it to 0).  This will
   subsequently cause the frame to be marked as SPEECH_BAD, if the FT of
   the frame is 0..7 for AMR or 0..8 for AMR-WB, or SID_BAD if the FT of
   the frame is 8 for AMR or 9 for AMR-WB, before it is passed to the
   speech decoder.  See [6] and [7] more details.




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   The following example shows an octet-aligned ToC with a CRC list for
   a payload containing 3 speech frames from a single channel session
   (assuming none of the FTs is equal to 14 or 15):

    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |1|  FT#1 |Q|P|P|1|  FT#2 |Q|P|P|0|  FT#3 |Q|P|P|     CRC#1     |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     CRC#2     |     CRC#3     |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   Each of the CRC's takes 8 bits

     0   1   2   3   4   5   6   7
   +---+---+---+---+---+---+---+---+
   | c0| c1| c2| c3| c4| c5| c6| c7|
   +---+---+---+---+---+---+---+---+

   and is calculated by the cyclic generator polynomial,

     C(x) = 1 + x^2 + x^3 + x^4 + x^8

   where ^ is the exponentiation operator.

   In binary form the polynomial has the following form: 101110001
   (MSB..LSB).

   The actual calculation of the CRC is made as follows:  First, an 8-
   bit CRC register is reset to zero: 00000000.  For each bit over which
   the CRC shall be calculated, an XOR operation is made between the
   rightmost bit of the CRC register and the bit.  The CRC register is
   then right shifted one step (inputting a "0" as the leftmost bit).
   If the result of the XOR operation mentioned above is a "1"
   "10111000" is then bit-wise XOR-ed into the CRC register.  This
   operation is repeated for each bit that the CRC should cover.  In
   this case, the first bit would be d(0) for the speech frame for which
   the CRC should cover.  When the last bit (e.g., d(54) for AMR 5.9
   according to Table 1 in Section 3.6) have been used in this CRC
   calculation, the contents in CRC register should simply be copied to
   the corresponding field in the list of CRC's.

   Fast calculation of the CRC on a general-purpose CPU is possible
   using a table-driven algorithm.







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4.4.3. Speech Data

   In octet-aligned mode, speech data is carried in a similar way to
   that in the bandwidth-efficient mode as discussed in Section 4.3.3,
   with the following exceptions:

      -  The last octet of each speech frame MUST be padded with zeroes
         at the end if not all bits in the octet are used.  In other
         words, each speech frame MUST be octet-aligned.

      -  When multiple speech frames are present in the speech data
         (i.e., compound payload), the speech frames can be arranged
         either one whole frame after another as usual, or with the
         octets of all frames interleaved together at the octet level.
         Since the bits within each frame are ordered with the most
         error-sensitive bits first, interleaving the octets collects
         those sensitive bits from all frames to be nearer the beginning
         of the packet.  This is called "robust sorting order" which
         allows the application of UED (such as UDP-Lite [15]) or UEP
         (such as the ULP [18]) mechanisms to the payload data.  The
         details of assembling the payload are given in the next
         section.

   The use of robust sorting order for a session MUST be agreed via
   out-of-band means.  Section 8 specifies a MIME parameter for this
   purpose.

   Note, robust sorting order MUST only be performed on the frame level
   and thus is independent of interleaving which is at the frame-block
   level, as described in Section 4.4.1. In other words, robust sorting
   can be applied to either non-interleaved or interleaved sessions.

4.4.4. Methods for Forming the Payload

   Two different packetization methods, namely normal order and robust
   sorting order, exist for forming a payload in octet-aligned mode.  In
   both cases, the payload header and table of contents are packed into
   the payload the same way; the difference is in the packing of the
   speech frames.

   The payload begins with the payload header of one octet or two if
   frame interleaving is selected.  The payload header is followed by
   the table of contents consisting of a list of one-octet ToC entries.
   If frame CRCs are to be included, they follow the table of contents
   with one 8-bit CRC filling each octet.  Note that if a given frame
   has a ToC entry with FT=14 or 15, there will be no CRC present.





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   The speech data follows the table of contents, or the CRCs if
   present.  For packetization in the normal order, all of the octets
   comprising a speech frame are appended to the payload as a unit. The
   speech frames are packed in the same order as their corresponding ToC
   entries are arranged in the ToC list, with the exception that if a
   given frame has a ToC entry with FT=14 or 15, there will be no data
   octets present for that frame.

   For packetization in robust sorting order, the octets of all speech
   frames are interleaved together at the octet level.  That is, the
   data portion of the payload begins with the first octet of the first
   frame, followed by the first octet of the second frame, then the
   first octet of the third frame, and so on.  After the first octet of
   the last frame has been appended, the cycle repeats with the second
   octet of each frame.  The process continues for as many octets as are
   present in the longest frame.  If the frames are not all the same
   octet length, a shorter frame is skipped once all octets in it have
   been appended.  The order of the frames in the cycle will be
   sequential if frame interleaving is not in use, or according to the
   interleave pattern specified in the payload header if frame
   interleaving is in use.  Note that if a given frame has a ToC entry
   with FT=14 or 15, there will be no data octets present for that frame
   so that frame is skipped in the robust sorting cycle.

   The UED and/or UEP is RECOMMENDED to cover at least the RTP header,
   payload header, table of contents, and class A bits of a sorted
   payload.  Exactly how many octets need to be covered depends on the
   network and application.  If CRCs are used together with robust
   sorting, only the RTP header, the payload header, and the ToC SHOULD
   be covered by UED/UEP.  The means to communicate to other layers
   performing UED/UEP the number of octets to be covered is beyond the
   scope of this specification.



















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4.4.5. Payload Examples

4.4.5.1. Basic Single Channel Payload Carrying Multiple Frames

   The following diagram shows an octet aligned payload from a single
   channel session that carries two AMR frames of 7.95 kbps coding mode
   (FT=5).  In the payload, a codec mode request is sent (CMR=6),
   requesting the encoder at the receiver's side to use AMR 10.2 kbps
   coding mode.  No frame CRC, interleaving, or robust-sorting is in
   use.

    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | CMR=6 |R|R|R|R|1|FT#1=5 |Q|P|P|0|FT#2=5 |Q|P|P|   f1(0..7)    |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   f1(8..15)   |  f1(16..23)   |  ....                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   : ...                                                           :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         ...   |f1(152..158) |P|   f2(0..7)    |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   f2(8..15)   |  f2(16..23)   |  ....                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   : ...                                                           :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         ...   |f2(152..158) |P|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   Note, in above example the last octet in both speech frames is padded
   with one 0 to make it octet-aligned.

4.4.5.2. Two Channel Payload with CRC, Interleaving, and Robust-sorting

   This example shows an octet aligned payload from a two channel
   session.  Two frame-blocks, each containing 2 speech frames of 7.95
   kbps coding mode (FT=5), are carried in this payload,

   The two channels are left (L) and right (R) with L coming before R.
   In the payload, a codec mode request is also sent (CMR=6), requesting
   the encoder at the receiver's side to use AMR 10.2 kbps coding mode.

   Moreover, frame CRC and frame-block interleaving are both enabled for
   the session.  The interleaving length is 2 (ILL=1) and this payload
   is the first one in an interleave group (ILP=0).

   The first two frames in the payload are the L and R channel speech
   frames of frame-block #1, consisting of bits f1L(0..158) and



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   f1R(0..158), respectively.  The next two frames are the L and R
   channel frames of frame-block #3, consisting of bits f3L(0..158) and
   f3R(0..158), respectively, due to interleaving.  For each of the four
   speech frames a CRC is calculated as CRC1L(0..7), CRC1R(0..7),
   CRC3L(0..7), and CRC3R(0..7), respectively.  Finally, the payload is
   robust sorted.

    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | CMR=6 |R|R|R|R| ILL=1 | ILP=0 |1|FT#1L=5|Q|P|P|1|FT#1R=5|Q|P|P|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |1|FT#3L=5|Q|P|P|0|FT#3R=5|Q|P|P|      CRC1L    |      CRC1R    |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |      CRC3L    |      CRC3R    |   f1L(0..7)   |   f1R(0..7)   |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   f3L(0..7)   |   f3R(0..7)   |  f1L(8..15)   |  f1R(8..15)   |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |  f3L(8..15)   |  f3R(8..15)   |  f1L(16..23)  |  f1R(16..23)  |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   : ...                                                           :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | f3L(144..151) | f3R(144..151) |f1L(152..158)|P|f1R(152..158)|P|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |f3L(152..158)|P|f3R(152..158)|P|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   Note, in above example the last octet in all the four speech frames
   is padded with one zero bit to make it octet-aligned.

4.5. Implementation Considerations

   An application implementing this payload format MUST understand all
   the payload parameters in the out-of-band signaling used.  For
   example, if an application uses SDP, all the SDP and MIME parameters
   in this document MUST be understood.  This requirement ensures that
   an implementation always can decide if it is capable or not of
   communicating.

   No operation mode of the payload format is mandatory to implement.
   The requirements of the application using the payload format should
   be used to determine what to implement.  To achieve basic
   interoperability an implementation SHOULD at least implement both
   bandwidth-efficient and octet-aligned mode for single channel.  The
   other operations mode: interleaving, robust sorting, frame-wise CRC
   in both single and multi-channel is OPTIONAL to implement.





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5. AMR and AMR-WB Storage Format

   The storage format is used for storing AMR or AMR-WB speech frames in
   a file or as an e-mail attachment.  Multiple channel content is
   supported.

   In general, an AMR or AMR-WB file has the following structure:

   +------------------+
   | Header           |
   +------------------+
   | Speech frame 1   |
   +------------------+
   : ...              :
   +------------------+
   | Speech frame n   |
   +------------------+

   Note, to preserve interoperability with already deployed
   implementations, single channel content uses a file header format
   different from that of multi-channel content.

5.1. Single channel Header

   A single channel AMR or AMR-WB file header contains only a magic
   number and different magic numbers are defined to distinguish AMR
   from AMR-WB.

   The magic number for single channel AMR files MUST consist of ASCII
   character string:

      "#!AMR\n"
      (or 0x2321414d520a in hexadecimal).

   The magic number for single channel AMR-WB files MUST consist of
   ASCII character string:

      "#!AMR-WB\n"
      (or 0x2321414d522d57420a in hexadecimal).

   Note, the "\n" is an important part of the magic numbers and MUST be
   included in the comparison, since, otherwise, the single channel
   magic numbers above will become indistinguishable from those of the
   multi-channel files defined in the next section.







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5.2. Multi-channel Header

   The multi-channel header consists of a magic number followed by a 32
   bit channel description field, giving the multi-channel header the
   following structure:

   +------------------+
   | magic number     |
   +------------------+
   | chan-desc field  |
   +------------------+

   The magic number for multi-channel AMR files MUST consist of the
   ASCII character string:

      "#!AMR_MC1.0\n"
      (or 0x2321414d525F4D43312E300a in hexadecimal).

   The magic number for multi-channel AMR-WB files MUST consist of the
   ASCII character string:

      "#!AMR-WB_MC1.0\n"
      (or 0x2321414d522d57425F4D43312E300a in hexadecimal).

   The version number in the magic numbers refers to the version of the
   file format.

   The 32 bit channel description field is defined as:

    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |      Reserved bits                                    | CHAN  |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   Reserved bits: MUST be set to 0 when written, and a reader MUST
                  ignore them.

   CHAN (4 bit unsigned integer): Indicates the number of audio channels
      contained in this storage file.  The valid values and the order of
      the channels within a frame block are specified in Section 4.1 in
      [24].









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5.3. Speech Frames

   After the file header, speech frame-blocks consecutive in time are
   stored in the file.  Each frame-block contains a number of octet-
   aligned speech frames equal to the number of channels, and stored in
   increasing order, starting with channel 1.

   Each stored speech frame starts with a one octet frame header with
   the following format:

    0 1 2 3 4 5 6 7
   +-+-+-+-+-+-+-+-+
   |P|  FT   |Q|P|P|
   +-+-+-+-+-+-+-+-+

   The FT field and the Q bit are defined in the same way as in Section
   4.1.2. The P bits are padding and MUST be set to 0.

   Following this one octet header come the speech bits as defined in
   4.3.3.  The last octet of each frame is padded with zeroes, if
   needed, to achieve octet alignment.

   The following example shows an AMR frame in 5.9 kbit coding mode
   (with 118 speech bits) in the storage 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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |P| FT=2  |Q|P|P|                                               |
   +-+-+-+-+-+-+-+-+                                               +
   |                                                               |
   +          Speech bits for frame-block n, channel k             +
   |                                                               |
   +                                                           +-+-+
   |                                                           |P|P|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   Frame-blocks or speech frames lost in transmission and non-received
   frame-blocks between SID updates during non-speech periods MUST be
   stored as NO_DATA frames (frame type 15, as defined in [2] and [4])
   or SPEECH_LOST (frame type 14, only available for AMR-WB) in complete
   frame-blocks to keep synchronization with the original media.









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6. Congestion Control

   The general congestion control considerations for transporting RTP
   data apply to AMR or AMR-WB speech over RTP as well.  However, the
   multi-rate capability of AMR and AMR-WB speech coding may provide an
   advantage over other payload formats for controlling congestion since
   the bandwidth demand can be adjusted by selecting a different coding
   mode.

   Another parameter that may impact the bandwidth demand for AMR and
   AMR-WB is the number of frame-blocks that are encapsulated in each
   RTP payload.  Packing more frame-blocks in each RTP payload can
   reduce the number of packets sent and hence the overhead from
   IP/UDP/RTP headers, at the expense of increased delay.

   If forward error correction (FEC) is used to combat packet loss, the
   amount of redundancy added by FEC will need to be regulated so that
   the use of FEC itself does not cause a congestion problem.

   It is RECOMMENDED that AMR or AMR-WB applications using this payload
   format employ congestion control.  The actual mechanism for
   congestion control is not specified but should be suitable for real-
   time flows, e.g., "Equation-Based Congestion Control for Unicast
   Applications" [17].

7. Security Considerations

   RTP packets using the payload format defined in this specification
   are subject to the general security considerations discussed in [8].

   As this format transports encoded speech, the main security issues
   include confidentiality and authentication of the speech itself.  The
   payload format itself does not have any built-in security mechanisms.
   External mechanisms, such as SRTP [22], MAY be used.

   This payload format does not exhibit any significant non-uniformity
   in the receiver side computational complexity for packet processing
   and thus is unlikely to pose a denial-of-service threat due to the
   receipt of pathological data.

7.1. Confidentiality

   To achieve confidentiality of the encoded AMR or AMR-WB speech, all
   speech data bits will need to be encrypted.  There is less a need to
   encrypt the payload header or the table of contents due to 1) that
   they only carry information about the requested speech mode, frame
   type, and frame quality, and 2) that this information could be useful
   to some third party, e.g., quality monitoring.



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   As long as the AMR or AMR-WB payload is only packed and unpacked at
   either end, encryption may be performed after packet encapsulation so
   that there is no conflict between the two operations.

   Interleaving may affect encryption.  Depending on the encryption
   scheme used, there may be restrictions on, for example, the time when
   keys can be changed.  Specifically, the key change may need to occur
   at the boundary between interleave groups.

   The type of encryption method used may impact the error robustness of
   the payload data.  The error robustness may be severely reduced when
   the data is encrypted unless an encryption method without error-
   propagation is used, e.g., a stream cipher.  Therefore, UED/UEP based
   on robust sorting may be difficult to apply when the payload data is
   encrypted.

7.2. Authentication

   To authenticate the sender of the speech, an external mechanism has
   to be used.  It is RECOMMENDED that such a mechanism protect all the
   speech data bits.  Note that the use of UED/UEP may be difficult to
   combine with authentication because any bit errors will cause
   authentication to fail.

   Data tampering by a man-in-the-middle attacker could result in
   erroneous depacketization/decoding that could lower the speech
   quality.  Tampering with the CMR field may result in speech in a
   different quality than desired.

   To prevent a man-in-the-middle attacker from tampering with the
   payload packets, some additional information besides the speech bits
   SHOULD be protected.  This may include the payload header, ToC, frame
   CRCs, RTP timestamp, RTP sequence number, and the RTP marker bit.

7.3. Decoding Validation

   When processing a received payload packet, if the receiver finds that
   the calculated payload length, based on the information of the
   session and the values found in the payload header fields, does not
   match the size of the received packet, the receiver SHOULD discard
   the packet.  This is because decoding a packet that has errors in its
   length field could severely degrade the speech quality.

8. Payload Format Parameters

   This section defines the parameters that may be used to select
   optional features of the AMR and AMR-WB payload formats.  The
   parameters are defined here as part of the MIME subtype registrations



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   for the AMR and AMR-WB speech codecs.  A mapping of the parameters
   into the Session Description Protocol (SDP) [11] is also provided for
   those applications that use SDP.  Equivalent parameters could be
   defined elsewhere for use with control protocols that do not use MIME
   or SDP.

   Two separate MIME registrations are made, one for AMR and one for
   AMR-WB, because they are distinct encodings that must be
   distinguished by the MIME subtype.

   The data format and parameters are specified for both real-time
   transport in RTP and for storage type applications such as e-mail
   attachments.

8.1. AMR MIME Registration

   The MIME subtype for the Adaptive Multi-Rate (AMR) codec is allocated
   from the IETF tree since AMR is expected to be a widely used speech
   codec in general VoIP applications.  This MIME registration covers
   both real-time transfer via RTP and non-real-time transfers via
   stored files.

   Note, any unspecified parameter MUST be ignored by the receiver.

   Media Type name:     audio

   Media subtype name:  AMR

   Required parameters: none

   Optional parameters:
      These parameters apply to RTP transfer only.

      octet-align: Permissible values are 0 and 1.  If 1, octet-aligned
               operation SHALL be used.  If 0 or if not present,
               bandwidth efficient operation is employed.

      mode-set:  Requested AMR mode set.  Restricts the active codec
               mode set to a subset of all modes.  Possible values are a
               comma separated list of modes from the set: 0,...,7 (see
               Table 1a [2]).  If such mode set is specified by the
               decoder, the encoder MUST abide by the request and MUST
               NOT use modes outside of the subset.  If not present, all
               codec modes are allowed for the session.

      mode-change-period: Specifies a number of frame-blocks, N, that is
               the interval at which codec mode changes are allowed.
               The initial phase of the interval is arbitrary, but



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               changes must be separated by multiples of N frame-blocks.
               If this parameter is not present, mode changes are
               allowed at any time during the session.

      mode-change-neighbor: Permissible values are 0 and 1.  If 1, mode
               changes SHALL only be made to the neighboring modes in
               the active codec mode set.  Neighboring modes are the
               ones closest in bit rate to the current mode, either the
               next higher or next lower rate.  If 0 or if not present,
               change between any two modes in the active codec mode set
               is allowed.

      maxptime:  The maximum amount of media which can be encapsulated
               in a payload packet, expressed as time in milliseconds.
               The time is calculated as the sum of the time the media
               present in the packet represents.  The time SHOULD be a
               multiple of the frame size.  If this parameter is not
               present, the sender MAY encapsulate any number of speech
               frames into one RTP packet.

      crc:  Permissible values are 0 and 1.  If 1, frame CRCs SHALL be
               included in the payload, otherwise not.  If crc=1, this
               also implies automatically that octet-aligned operation
               SHALL be used for the session.

      robust-sorting: Permissible values are 0 and 1.  If 1, the payload
               SHALL employ robust payload sorting.  If 0 or if not
               present, simple payload sorting SHALL be used.  If
               robust-sorting=1, this also implies automatically that
               octet-aligned operation SHALL be used for the session.

      interleaving: Indicates that frame-block level interleaving SHALL
               be used for the session and its value defines the maximum
               number of frame-blocks allowed in an interleaving group
               (see Section 4.4.1).  If this parameter is not present,
               interleaving SHALL not be used.  The presence of this
               parameter also implies automatically that octet-aligned
               operation SHALL be used.

      ptime:     see RFC2327 [11].

      channels: The number of audio channels.  The possible values and
               their respective channel order is specified in section
               4.1 in [24].  If omitted it has the default value of 1.

   Encoding considerations:
               This type is defined for transfer via both RTP (RFC 1889)
               and stored-file methods as described in Sections 4 and 5,



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               respectively, of RFC 3267.  Audio data is binary data,
               and must be encoded for non-binary transport; the Base64
               encoding is suitable for Email.

   Security considerations:
               See Section 7 of RFC 3267.

   Public specification:
               Please refer to Section 11 of RFC 3267.

   Additional information:

               The following applies to stored-file transfer methods:

               Magic numbers:
                 single channel:
                 ASCII character string "#!AMR\n"
                 (or 0x2321414d520a in hexadecimal)
                 multi-channel:
                 ASCII character string "#!AMR_MC1.0\n"
                 (or 0x2321414d525F4D43312E300a in hexadecimal)

   File extensions: amr, AMR
   Macintosh file type code: none
   Object identifier or OID: none

   Person & email address to contact for further information:
               johan.sjoberg@ericsson.com
               ari.lakaniemi@nokia.com

   Intended usage: COMMON.
               It is expected that many VoIP applications (as well as
               mobile applications) will use this type.

   Author/Change controller:
               johan.sjoberg@ericsson.com
               ari.lakaniemi@nokia.com
               IETF Audio/Video transport working group

8.2. AMR-WB MIME Registration

   The MIME subtype for the Adaptive Multi-Rate Wideband (AMR-WB) codec
   is allocated from the IETF tree since AMR-WB is expected to be a
   widely used speech codec in general VoIP applications.  This MIME
   registration covers both real-time transfer via RTP and non-real-time
   transfers via stored files.

   Note, any unspecified parameter MUST be ignored by the receiver.



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   Media Type name:     audio

   Media subtype name:  AMR-WB

   Required parameters: none

   Optional parameters:

      These parameters apply to RTP transfer only.

      octet-align: Permissible values are 0 and 1.  If 1, octet-aligned
               operation SHALL be used.  If 0 or if not present,
               bandwidth efficient operation is employed.

      mode-set:  Requested AMR-WB mode set.  Restricts the active codec
               mode set to a subset of all modes.  Possible values are a
               comma separated list of modes from the set: 0,...,8 (see
               Table 1a [4]).  If such mode set is specified by the
               decoder, the encoder MUST abide by the request and MUST
               NOT use modes outside of the subset.  If not present, all
               codec modes are allowed for the session.

      mode-change-period: Specifies a number of frame-blocks, N, that is
               the interval at which codec mode changes are allowed.
               The initial phase of the interval is arbitrary, but
               changes must be separated by multiples of N frame-blocks.
               If this parameter is not present, mode changes are
               allowed at any time during the session.

      mode-change-neighbor: Permissible values are 0 and 1.  If 1, mode
               changes SHALL only be made to the neighboring modes in
               the active codec mode set.  Neighboring modes are the
               ones closest in bit rate to the current mode, either the
               next higher or next lower rate.  If 0 or if not present,
               change between any two modes in the active codec mode set
               is allowed.

      maxptime:  The maximum amount of media which can be encapsulated
               in a payload packet, expressed as time in milliseconds.
               The time is calculated as the sum of the time the media
               present in the packet represents.  The time SHOULD be a
               multiple of the frame size.  If this parameter is not
               present, the sender MAY encapsulate any number of speech
               frames into one RTP packet.







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      crc:  Permissible values are 0 and 1.  If 1, frame CRCs SHALL be
               included in the payload, otherwise not.  If crc=1, this
               also implies automatically that octet-aligned operation
               SHALL be used for the session.

      robust-sorting: Permissible values are 0 and 1.  If 1, the payload
               SHALL employ robust payload sorting.  If 0 or if not
               present, simple payload sorting SHALL be used.  If
               robust-sorting=1, this also implies automatically that
               octet-aligned operation SHALL be used for the session.

      interleaving: Indicates that frame-block level interleaving SHALL
               be used for the session and its value defines the maximum
               number of frame-blocks allowed in an interleaving group
               (see Section 4.4.1).  If this parameter is not present,
               interleaving SHALL not be used.  The presence of this
               parameter also implies automatically that octet-aligned
               operation SHALL be used.

      ptime:     see RFC2327 [11].

      channels: The number of audio channels.  The possible values and
               their respective channel order is specified in section
               4.1 in [24].  If omitted it has the default value of 1.

   Encoding considerations:
               This type is defined for transfer via both RTP (RFC 1889)
               and stored-file methods as described in Sections 4 and 5,
               respectively, of RFC 3267.  Audio data is binary data,
               and must be encoded for non-binary transport; the Base64
               encoding is suitable for Email.

   Security considerations:
               See Section 7 of RFC 3267.

   Public specification:
               Please refer to Section 11 of RFC 3267.

   Additional information:
               The following applies to stored-file transfer methods:

               Magic numbers:
                 single channel:
                 ASCII character string "#!AMR-WB\n"
                 (or 0x2321414d522d57420a in hexadecimal)
                 multi-channel:
                 ASCII character string "#!AMR-WB_MC1.0\n"
                 (or 0x2321414d522d57425F4D43312E300a in hexadecimal)



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               File extensions: awb, AWB
               Macintosh file type code: none
               Object identifier or OID: none

   Person & email address to contact for further information:
               johan.sjoberg@ericsson.com
               ari.lakaniemi@nokia.com

   Intended usage: COMMON.
               It is expected that many VoIP applications (as well as
               mobile applications) will use this type.

   Author/Change controller:
               johan.sjoberg@ericsson.com
               ari.lakaniemi@nokia.com
               IETF Audio/Video transport working group

8.3. Mapping MIME Parameters into SDP

   The information carried in the MIME media type specification has a
   specific mapping to fields in the Session Description Protocol (SDP)
   [11], which is commonly used to describe RTP sessions.  When SDP is
   used to specify sessions employing the AMR or AMR-WB codec, the
   mapping is as follows:

      -  The MIME type ("audio") goes in SDP "m=" as the media name.

      -  The MIME subtype (payload format name) goes in SDP "a=rtpmap"
         as the encoding name.  The RTP clock rate in "a=rtpmap" MUST be
         8000 for AMR and 16000 for AMR-WB, and the encoding parameters
         (number of channels) MUST either be explicitly set to N or
         omitted, implying a default value of 1.  The values of N that
         are allowed is specified in Section 4.1 in [24].

      -  The parameters "ptime" and "maxptime" go in the SDP "a=ptime"
         and "a=maxptime" attributes, respectively.

      -  Any remaining parameters go in the SDP "a=fmtp" attribute by
         copying them directly from the MIME media type string as a
         semicolon separated list of parameter=value pairs.

   Some example SDP session descriptions utilizing AMR and AMR-WB
   encodings follow.  In these examples, long a=fmtp lines are folded to
   meet the column width constraints of this document; the backslash
   ("\") at the end of a line and the carriage return that follows it
   should be ignored.





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   Example of usage of AMR in a possible GSM gateway scenario:

    m=audio 49120 RTP/AVP 97
    a=rtpmap:97 AMR/8000/1
    a=fmtp:97 mode-set=0,2,5,7; mode-change-period=2; \
      mode-change-neighbor=1
    a=maxptime:20

   Example of usage of AMR-WB in a possible VoIP scenario:

    m=audio 49120 RTP/AVP 98
    a=rtpmap:98 AMR-WB/16000
    a=fmtp:98 octet-align=1

   Example of usage of AMR-WB in a possible streaming scenario (two
   channel stereo):

    m=audio 49120 RTP/AVP 99
    a=rtpmap:99 AMR-WB/16000/2
    a=fmtp:99 interleaving=30
    a=maxptime:100

   Note that the payload format (encoding) names are commonly shown in
   upper case.  MIME subtypes are commonly shown in lower case.  These
   names are case-insensitive in both places.  Similarly, parameter
   names are case-insensitive both in MIME types and in the default
   mapping to the SDP a=fmtp attribute.

9. IANA Considerations

   Two new MIME subtypes have been registered, see Section 8.  A new SDP
   attribute "maxptime", defined in Section 8, has also been registered.
   The "maxptime" attribute is expected to be defined in the revision of
   RFC 2327 [11] and is added here with a consistent definition.

10.  Acknowledgements

   The authors would like to thank Petri Koskelainen, Bernhard Wimmer,
   Tim Fingscheidt, Sanjay Gupta, Stephen Casner, and Colin Perkins for
   their significant contributions made throughout the writing and
   reviewing of this document.

11.  References

   [1]   3GPP TS 26.090, "Adaptive Multi-Rate (AMR) speech transcoding",
         version 4.0.0 (2001-03), 3rd Generation Partnership Project
         (3GPP).




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   [2]   3GPP TS 26.101, "AMR Speech Codec Frame Structure", version
         4.1.0 (2001-06), 3rd Generation Partnership Project (3GPP).

   [3]   3GPP TS 26.190 "AMR Wideband speech codec; Transcoding
         functions", version 5.0.0 (2001-03), 3rd Generation Partnership
         Project (3GPP).

   [4]   3GPP TS 26.201 "AMR Wideband speech codec; Frame Structure",
         version 5.0.0 (2001-03), 3rd Generation Partnership Project
         (3GPP).

   [5]   Bradner, S., "Key words for use in RFCs to Indicate Requirement
         Levels", BCP 14, RFC 2119, March 1997.

   [6]   3GPP TS 26.093, "AMR Speech Codec; Source Controlled Rate
         operation", version 4.0.0 (2000-12), 3rd Generation Partnership
         Project (3GPP).

   [7]   3GPP TS 26.193 "AMR Wideband Speech Codec; Source Controlled
         Rate operation", version 5.0.0 (2001-03), 3rd Generation
         Partnership Project (3GPP).

   [8]   Schulzrinne, H, Casner, S., Frederick, R. and V. Jacobson,
         "RTP: A Transport Protocol for Real-Time Applications", RFC
         1889, January 1996.

   [9]   3GPP TS 26.092, "AMR Speech Codec; Comfort noise aspects",
         version 4.0.0 (2001-03), 3rd Generation Partnership Project
         (3GPP).

   [10]  3GPP TS 26.192 "AMR Wideband speech codec; Comfort Noise
         aspects", version 5.0.0 (2001-03), 3rd Generation Partnership
         Project (3GPP).

   [11]  Handley, M. and V. Jacobson, "SDP: Session Description
         Protocol", RFC 2327, April 1998.

   [24]  Schulzrinne, H., "RTP Profile for Audio and Video Conferences
         with Minimal Control" RFC 1890, January 1996.

11.1 Informative References

   [12]  GSM 06.60, "Enhanced Full Rate (EFR) speech transcoding",
         version 8.0.1 (2000-11), European Telecommunications Standards
         Institute (ETSI).






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   [13]  ANSI/TIA/EIA-136-Rev.C, part 410 - "TDMA Cellular/PCS - Radio
         Interface, Enhanced Full Rate Voice Codec (ACELP)." Formerly
         IS-641.  TIA published standard, June 1 2001.

   [14]  ARIB, RCR STD-27H, "Personal Digital Cellular Telecommunication
         System RCR Standard", Association of Radio Industries and
         Businesses (ARIB).

   [15]  Larzon, L., Degermark, M. and S. Pink, "The UDP Lite Protocol",
         Work in Progress.

   [16]  3GPP TS 25.415 "UTRAN Iu Interface User Plane Protocols",
         version 4.2.0 (2001-09), 3rd Generation Partnership Project
         (3GPP).

   [17]  S. Floyd, M. Handley, J. Padhye, J. Widmer, "Equation-Based
         Congestion Control for Unicast Applications", ACM SIGCOMM 2000,
         Stockholm, Sweden .

   [18]  Li, A., et. al., "An RTP Payload Format for Generic FEC with
         Uneven Level Protection", Work in Progress.

   [19]  Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format for
         Generic Forward Error Correction", RFC 2733, December 1999.

   [20]  3GPP TS 26.102, "AMR speech codec interface to Iu and Uu",
         version 4.0.0 (2001-03), 3rd Generation Partnership Project
         (3GPP).

   [21]  3GPP TS 26.202 "AMR Wideband speech codec; Interface to Iu and
         Uu", version 5.0.0 (2001-03), 3rd Generation Partnership
         Project (3GPP).

   [22]  Baugher, et. al., "The Secure Real Time Transport Protocol",
         Work in Progress.

   [23]  Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., Handley,
         M., Bolot, J., Vega-Garcia, A. and S. Fosse-Parisis, "RTP
         Payload for Redundant Audio Data", RFC 2198, September 1997.

   ETSI documents can be downloaded from the ETSI web server,
   "http://www.etsi.org/".  Any 3GPP document can be downloaded from the
   3GPP webserver, "http://www.3gpp.org/", see specifications.  TIA
   documents can be obtained from "www.tiaonline.org".







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12.  Authors' Addresses

   Johan Sjoberg
   Ericsson Research
   Ericsson AB
   SE-164 80 Stockholm, SWEDEN

   Phone:   +46 8 50878230
   EMail: Johan.Sjoberg@ericsson.com


   Magnus Westerlund
   Ericsson Research
   Ericsson AB
   SE-164 80 Stockholm, SWEDEN

   Phone:   +46 8 4048287
   EMail: Magnus.Westerlund@ericsson.com


   Ari Lakaniemi
   Nokia Research Center
   P.O.Box 407
   FIN-00045 Nokia Group, FINLAND

   Phone:   +358-71-8008000
   EMail: ari.lakaniemi@nokia.com


   Qiaobing Xie
   Motorola, Inc.
   1501 W. Shure Drive, 2-B8
   Arlington Heights, IL 60004, USA

   Phone:   +1-847-632-3028
   EMail: qxie1@email.mot.com















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13.  Full Copyright Statement

   Copyright (C) The Internet Society (2002).  All Rights Reserved.

   This document and translations of it may be copied and furnished to
   others, and derivative works that comment on or otherwise explain it
   or assist in its implementation may be prepared, copied, published
   and distributed, in whole or in part, without restriction of any
   kind, provided that the above copyright notice and this paragraph are
   included on all such copies and derivative works.  However, this
   document itself may not be modified in any way, such as by removing
   the copyright notice or references to the Internet Society or other
   Internet organizations, except as needed for the purpose of
   developing Internet standards in which case the procedures for
   copyrights defined in the Internet Standards process must be
   followed, or as required to translate it into languages other than
   English.

   The limited permissions granted above are perpetual and will not be
   revoked by the Internet Society or its successors or assigns.

   This document and the information contained herein is provided on an
   "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
   TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
   BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
   HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
   MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.

Acknowledgement

   Funding for the RFC Editor function is currently provided by the
   Internet Society.



















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