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Versions: (draft-sjoberg-avt-rtp-amr) 00 01 02 03 04 05 06 07 08 09 10 11 12 RFC 3267

Internet Engineering Task Force                  Johan Sjoberg, Ericsson
Audio Video Transport WG                     Magnus Westerlund, Ericsson
INTERNET-DRAFT                                      Ari Lakaniemi, Nokia
February 14, 2002                                  Qiaobing Xie, Motorola
Expires: August 13, 2002



  RTP payload format and file storage format for AMR and AMR-WB audio
                    <draft-ietf-avt-rtp-amr-12.txt>


Status of this Memo


   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups. Note that other
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   Internet-Drafts are draft documents valid for a maximum of six months
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   time. It is inappropriate to use Internet-Drafts as reference
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   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/lid-abstracts.txt

   The list of Internet-Draft Shadow Directories can be accessed at
   http://www.ietf.org/shadow.html

   This document is an individual submission to the IETF. Comments
   should be directed to the authors.



Abstract

   This document specifies a real-time transport protocol (RTP)
   payload format to be used for AMR and 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.


Table of Contents

   1. Introduction.................................................3

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   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...4
     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...............6
       3.6.1. Applying UEP and UED in an IP Network................7
     3.7. Robustness against Packet Loss...........................8
       3.7.1. Use of Forward Error Correction (FEC)................8
       3.7.2. Use of Frame Interleaving............................10
     3.8. Bandwidth Efficient or Octet-aligned Mode................10
     3.9. AMR or AMR-WB Speech over IP scenarios...................11
   4. AMR and AMR-WB RTP Payload Formats...........................13
     4.1. RTP Header Usage.........................................13
     4.2. Payload Structure........................................14
     4.3. Bandwidth-Efficient Mode.................................14
       4.3.1. The Payload Header...................................14
       4.3.2. The Payload Table of Contents........................16
       4.3.3. Speech Data..........................................17
       4.3.4. Algorithm for Forming the Payload....................18
       4.3.5 Payload Examples......................................18
         4.3.5.1. Single Channel Payload Carrying a Single Frame...18
         4.3.5.2. Single Channel Payload Carrying Multiple Frames..19
         4.3.5.3. Multi-Channel Payload Carrying Multiple Frames...20
     4.4. Octet-aligned Mode.......................................21
       4.4.1. The Payload Header...................................21
       4.4.2. The Payload Table of Contents and Frame CRCs.........22
         4.4.2.1. Use of Frame CRC for UED over IP.................24
       4.4.3. Speech Data..........................................25
       4.4.4. Methods for Forming the Payload......................26
       4.4.5. Payload Examples.....................................27
         4.4.5.1. Basic Single Channel Payload Carrying
                  Multiple Frames..................................27
         4.4.5.2. Two Channel Payload with CRC, Interleaving,
                  and Robust-sorting...............................27
     4.5. Implementation Considerations............................28
   5. AMR and AMR-WB Storage Format................................28
     5.1. Single Channel Header....................................29
     5.2. Multi-channel Header.....................................29
     5.3. Speech Frames............................................30
   6. Congestion Control...........................................31
   7. Security Considerations......................................32
     7.1. Confidentiality..........................................32
     7.2. Authentication...........................................32
     7.3. Decoding Validation......................................33
   8. Payload Format Parameters....................................33
     8.1. AMR MIME Registration....................................33
     8.2. AMR-WB MIME Registration.................................36
     8.3. Mapping MIME Parameters into SDP.........................38
   9. IANA Considerations..........................................39
   10. Acknowledgements............................................39
   11. References..................................................39
     11.1 Informative References...................................40

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   12. Authors' Addresses..........................................41


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
    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.

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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.

   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

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   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 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.


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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, e.g., using the
   rtpmap attribute in SDP, 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 and order of channels
   carried in each frame-block are 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.

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                     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.

   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] (work in progress), 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

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   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.

   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

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   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.

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

   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-2)..p(n+3) 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.


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   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
   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 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.

    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

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   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.

   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

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       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.

   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

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   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).

   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  |
   +-+-+-+-+

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   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.

   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).


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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.

   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).

   The frame quality indicator is included for interoperability with
   the ATM payload format described in ITU-T I.366.2, the UMTS Iu

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   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.

   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

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   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).


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).

    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):
      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).

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   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.

   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.

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   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.

   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.



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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.

   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|
   +---+---+---+---+---+---+---+---+


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   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.


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

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   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.

   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

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   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.


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

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   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
   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.


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:

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   +------------------+
   | 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.


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.

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   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): Specifies the number and formation
     of audio channels contained in this storage file, as defined in
     the following table:

           |  # of   |              |         channel
      CHAN | channels| description  |  1   2   3   4   5   6
     ============================================================
       1   |  2      | stereo       |  l   r
       2   |  3      |              |  l   r   c
       3   |  4      | quadrophonic |  Fl  Fr  Rl  Rr
       4   |  4      |              |  l   c   r   S
       5   |  5      |              |  Fl  Fr  Fc  Sl  Sr
       6   |  6      |              |  l   lc  c   r   rc  S
     ------+-----------------------------------------------------
     0,7-15| Reserved for future use
     ============================================================
       Legends:
         l - left
         r - right
         c - center
         S - surround
         F - front
         R - rear

     Table 2: Channel definitions for the storage format


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|
   +-+-+-+-+-+-+-+-+


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   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.


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].


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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.

   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

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   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 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

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   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
               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

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INTERNET-DRAFT    RTP Payload Format for AMR and AMR-WB     January 2002

               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 XXXX. 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 XXXX.

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

   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



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INTERNET-DRAFT    RTP Payload Format for AMR and AMR-WB     January 2002


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.

   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.

    crc:       Permissible values are 0 and 1. If 1, frame CRCs SHALL
               be included in the payload, otherwise not. If crc=1,

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INTERNET-DRAFT    RTP Payload Format for AMR and AMR-WB     January 2002

               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 XXXX. 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 XXXX.

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

   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)

               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

Sjoberg et al.                                                 [Page 37]


INTERNET-DRAFT    RTP Payload Format for AMR and AMR-WB     January 2002

               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.

   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):

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    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 are to be registered, see Section 8. A new
   SDP attribute "maxptime", also defined in Section 8, needs to be
   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).

   [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]  S. Bradner, "Key words for use in RFCs to Indicate
        Requirement Levels", IETF 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).


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INTERNET-DRAFT    RTP Payload Format for AMR and AMR-WB     January 2002

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

   [9]  GSM 06.92, "Comfort noise aspects for Adaptive Multi-Rate
        (AMR) speech traffic channels", version 7.1.1 (1999-12),
        European Telecommunications Standards Institute (ETSI).

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

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

   [24] H. Schulzrinne, "RTP Profile for Audio and Video Conferences
        with Minimal Control" IETF 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).

   [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] Lars-Ake Larzon, Mikael Degermark, and Stephen Pink, "The UDP
        Lite Protocol", IETF Draft (Work in Progress), February 23,
        2001.

   [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] A. Li, et al., "An RTP Payload Format for Generic FEC with
        Uneven Level Protection ", IETF Draft (Work in Progress),
        October 2001.

   [19] J. Rosenberg, and H. Schulzrinne, "An RTP Payload Format

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INTERNET-DRAFT    RTP Payload Format for AMR and AMR-WB     January 2002

        for Generic Forward Error Correction", IETF 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",
        IETF Draft (Work in Progress), November 2001.

   [23] C. Perkins, et al., "RTP Payload for Redundant Audio Data",
        IETF 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".


12. Authors' Addresses

   Johan Sjoberg                  Tel:   +46 8 50878230
   Ericsson Research              EMail: Johan.Sjoberg@ericsson.com
   Ericsson Radio Systems AB
   SE-164 80 Stockholm, SWEDEN

   Magnus Westerlund              Tel:   +46 8 4048287
   Ericsson Research              EMail: Magnus.Westerlund@ericsson.com
   Ericsson Radio Systems AB
   SE-164 80 Stockholm, SWEDEN

   Ari Lakaniemi                  Tel:   +358-71-8008000
   Nokia Research Center          EMail: ari.lakaniemi@nokia.com
   P.O.Box 407
   FIN-00045 Nokia Group, FINLAND

   Qiaobing Xie                   Tel:   +1-847-632-3028
   Motorola, Inc.                 EMail: qxie1@email.mot.com
   1501 W. Shure Drive, 2-B8
   Arlington Heights, IL 60004, USA





   This Internet-Draft expires in six months from February 2002.





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