path: root/doc/rfc/rfc4867.txt

Network Working Group                                         J. Sjoberg
Request for Comments: 4867                                 M. Westerlund
Obsoletes: 3267                                                 Ericsson
Category: Standards Track                                   A. Lakaniemi
                                                                   Nokia
                                                                  Q. Xie
                                                                Motorola
                                                              April 2007


          RTP Payload Format and File Storage Format for the
  Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB)
                              Audio Codecs

Status of This Memo

   This document specifies an Internet standards track protocol for the
   Internet community, and requests discussion and suggestions for
   improvements.  Please refer to the current edition of the "Internet
   Official Protocol Standards" (STD 1) for the standardization state
   and status of this protocol.  Distribution of this memo is unlimited.

Copyright Notice

   Copyright (C) The IETF Trust (2007).

Abstract

   This document specifies a Real-time Transport Protocol (RTP) payload
   format to be used for Adaptive Multi-Rate (AMR) and Adaptive Multi-
   Rate Wideband (AMR-WB) encoded speech signals.  The payload format is
   designed to be able to interoperate with existing AMR and AMR-WB
   transport formats on non-IP networks.  In addition, a file format is
   specified for transport of AMR and AMR-WB speech data in storage mode
   applications such as email.  Two separate media type registrations
   are included, one for AMR and one for AMR-WB, specifying use of both
   the RTP payload format and the storage format.  This document
   obsoletes RFC 3267.













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 Table of Contents

   1. Introduction ....................................................4
   2. Conventions and Acronyms ........................................4
   3. Background on AMR/AMR-WB and Design Principles ..................5
      3.1. The Adaptive Multi-Rate (AMR) Speech Codec .................5
      3.2. The Adaptive Multi-Rate Wideband (AMR-WB) Speech Codec .....6
      3.3. Multi-Rate Encoding and Mode Adaptation ....................6
      3.4. Voice Activity Detection and Discontinuous Transmission ....7
      3.5. Support for Multi-Channel Session ..........................7
      3.6. Unequal Bit-Error Detection and Protection .................8
           3.6.1. Applying UEP and UED in an IP Network ...............8
      3.7. Robustness against Packet Loss ............................10
           3.7.1. Use of Forward Error Correction (FEC) ..............10
           3.7.2. Use of Frame Interleaving ..........................12
      3.8. Bandwidth-Efficient or Octet-Aligned Mode .................12
      3.9. AMR or AMR-WB Speech over IP Scenarios ....................13
   4. AMR and AMR-WB RTP Payload Formats .............................15
      4.1. RTP Header Usage ..........................................15
      4.2. Payload Structure .........................................17
      4.3. Bandwidth-Efficient Mode ..................................17
           4.3.1. The Payload Header .................................17
           4.3.2. The Payload Table of Contents ......................18
           4.3.3. Speech Data ........................................20
           4.3.4. Algorithm for Forming the Payload ..................21
           4.3.5. Payload Examples ...................................21
                  4.3.5.1. Single-Channel Payload Carrying a
                           Single Frame ..............................21
                  4.3.5.2. Single-Channel Payload Carrying
                           Multiple Frames ...........................22
                  4.3.5.3. Multi-Channel Payload Carrying
                           Multiple Frames ...........................23
      4.4. Octet-Aligned Mode ........................................25
           4.4.1. The Payload Header .................................25
           4.4.2. The Payload Table of Contents and Frame CRCs .......26
                  4.4.2.1. Use of Frame CRC for UED over IP ..........28
           4.4.3. Speech Data ........................................30
           4.4.4. Methods for Forming the Payload ....................31
           4.4.5. Payload Examples ...................................32
                  4.4.5.1. Basic Single-Channel Payload
                           Carrying Multiple Frames ..................32
                  4.4.5.2. Two-Channel Payload with CRC,
                           Interleaving, and Robust Sorting ..........32
      4.5. Implementation Considerations .............................33
           4.5.1. Decoding Validation ................................34
   5. AMR and AMR-WB Storage Format ..................................35
      5.1. Single-Channel Header .....................................35
      5.2. Multi-Channel Header ......................................36



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      5.3. Speech Frames .............................................37
   6. Congestion Control .............................................38
   7. Security Considerations ........................................38
      7.1. Confidentiality ...........................................39
      7.2. Authentication and Integrity ..............................39
   8. Payload Format Parameters ......................................39
      8.1. AMR Media Type Registration ...............................40
      8.2. AMR-WB Media Type Registration ............................44
      8.3. Mapping Media Type Parameters into SDP ....................47
           8.3.1. Offer-Answer Model Considerations ..................48
           8.3.2. Usage of Declarative SDP ...........................50
           8.3.3. Examples ...........................................51
   9. IANA Considerations ............................................53
   10. Changes from RFC 3267 .........................................53
   11. Acknowledgements ..............................................55
   12. References ....................................................55
      12.1. Normative References .....................................55
      12.2. Informative References ...................................56

































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1.  Introduction

   This document obsoletes RFC 3267 and extends that specification with
   offer/answer rules.  See Section 10 for the changes made to this
   format in relation to RFC 3267.

   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 media 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 RFC 2119 [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




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   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.

   The byte order used in this document is network byte order, i.e., the
   most significant byte first.  The bit order is also the most
   significant bit first.  This is presented in all figures as having
   the most significant bit leftmost on a line and with the lowest
   number.  Some bit fields may wrap over multiple lines in which cases
   the bits on the first line are more significant than the bits on the
   next line.

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 media 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 codec 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 eight narrow band
   speech encoding modes with bit rates between 4.75 and 12.2 kbps.  The
   sampling frequency used in AMR is 8000 Hz and the speech encoding is
   performed on 20 ms speech frames.  Therefore, each encoded AMR speech
   frame represents 160 samples of the original speech.




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   Among the eight AMR encoding modes, three are already separately
   adopted as standards of their own.  Particularly, the 6.7 kbps mode
   is adopted as PDC-EFR [18], the 7.4 kbps mode as IS-641 codec in TDMA
   [17], and the 12.2 kbps mode as GSM-EFR [16].

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 nine wide band speech coding modes with respective
   bit rates ranging from 6.6 to 23.85 kbps.  The sampling frequency
   used in AMR-WB is 16000 Hz and the speech processing is performed on
   20 ms frames.  This means that each AMR-WB encoded frame represents
   320 speech samples.

3.3.  Multi-Rate Encoding and Mode Adaptation

   The multi-rate encoding (i.e., multi-mode) capability of AMR and
   AMR-WB is designed for preserving high speech quality under a wide
   range of transmission conditions.

   With AMR or AMR-WB, mobile radio systems are able to use available
   bandwidth as effectively as possible.  For example, 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.  This enables the best possible
   trade-off between speech compression rate and error tolerance.  To
   perform mode adaptation, the decoder (speech receiver) needs to
   signal the encoder (speech sender) the new mode it prefers.  This
   mode change signal is called Codec Mode Request or CMR.

   Since in most sessions speech is sent in both directions between the
   two ends, the mode requests from the decoder at one end to the
   encoder at the other end are piggy-backed over the speech frames in
   the reverse direction.  In other words, there is no out-of-band
   signaling needed for sending CMRs.

   Every AMR or AMR-WB codec implementation is required to support all
   the respective speech coding modes defined by the codec and must be
   able to handle mode switching to any of the modes at any time.
   However, some transport systems may impose limitations in the number
   of modes supported and how often the mode can change due to bandwidth





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   limitations or other constraints.  For this reason, the decoder is
   allowed to indicate its acceptance of a particular mode or a subset
   of the defined modes for the session using out-of-band means.

   For example, the GSM radio link can only use a subset of at most four
   different modes in a given session.  This subset can be any
   combination of the eight AMR modes for an AMR session or any
   combination of the nine 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 media type parameters that may be used
   to signal these mode adaptation controls at session setup.

3.4.  Voice Activity Detection and Discontinuous Transmission

   Both codecs support voice activity detection (VAD) and generation of
   comfort noise (CN) parameters during silence periods.  Hence, the
   codecs have the option to reduce the number of transmitted bits and
   packets during silence periods to a minimum.  The operation of
   sending CN parameters at regular intervals during silence periods is
   usually called discontinuous transmission (DTX) or source controlled
   rate (SCR) operation.  The AMR or AMR-WB frames containing CN
   parameters are called Silence Indicator (SID) frames.  See more
   details about VAD and DTX functionality in [9] and [10].

3.5.  Support for Multi-Channel Session

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

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

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

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



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   implied by the number of channels as specified in Section 4.1 in
   [12].

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 the
   most sensitive and bits in class C the least sensitive (see Table 1
   below for AMR and [4] for AMR-WB).  An AMR or AMR-WB frame is only
   declared damaged if there are bit errors found in the most sensitive
   bits, i.e., the class A bits.  On the other hand, it is acceptable to
   have some bit errors in the other bits, i.e., class B and C bits.

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

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

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

3.6.1.  Applying UEP and UED in an IP Network

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




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

   Link-layer protocols exist that do not discard packets containing 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 [19]), bit error tolerant AMR and AMR-WB traffic could
   achieve better performance over these types of links.  The
   relationship between UDP-Lite's partial checksum at the transport
   layer and the checksum coverage provided by the link-layer frame is
   described in UDP-Lite specification [19].

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

   a) Utilizing a partial checksum to cover the IP, transport protocol
      (e.g., UDP-Lite), RTP and payload headers, and the most important
      speech bits of the payload.  The IP, UDP and RTP headers need to
      be protected, and it is recommended that at least all class A bits
      are covered by the checksum.

   b) Utilizing a partial checksum to only cover the IP, transport
      protocol, RTP and payload headers, but an AMR or AMR-WB frame CRC
      to cover the class A bits of each speech frame in the RTP payload.

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

      Note, it is still important that the network designer pays
      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 the Universal Mobile
      Telecommunications System (UMTS) can be found in [24] and for
      AMR-WB in [25].

   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.




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   Approach 1 is bit efficient, flexible and simple, 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
   AMR or AMR-WB frames in a RTP 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
   improves the speech quality when transporting multiple AMR or AMR-WB
   frames over links subject to bit errors.

   The choice between the above two approaches must be made based on the
   available bandwidth, and the desired tolerance to bit errors.
   Neither solution is appropriate for all cases.  Section 8 defines
   parameters that may be used at session setup to choose 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., RFC 2733 [23], which
   generates extra packets containing repair data.  The whole payload
   can also be sorted in sensitivity order to support external FEC
   schemes using UEP.  There is also a work in progress on a generic
   version of such a scheme [22] 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 a frame using either the
   same mode or another mode, e.g., one with lower bandwidth.  We
   describe such a scheme next.







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

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

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

              Figure 1: An example of redundant transmission

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

   The use of this approach does not require signaling at the session
   setup.  However, a parameter for providing a maximum delay in
   transmitting any redundant frame is defined in Section 8.  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" [27].
   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 the duration of 5 frames, the
   bandwidth overhead of RFC 2198 will be lower; or, if a non-AMR codec
   is desired for the redundant encoding, the AMR payload format won't
   be able to carry it.

   The sender is responsible for selecting an appropriate amount of
   redundancy based on feedback about the channel, e.g., in RTCP



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   receiver reports.  A sender should not base selection of FEC on the
   CMR, as this parameter most probably was set based on non-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 to be encapsulated into a single RTP packet.  One
   of the drawbacks of such an approach is that packet loss can cause
   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 more 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 is 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.

   A streaming service has less strict real-time requirements and
   therefore can use a larger number of frame-blocks per packet than a
   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 that 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 is when AMR or AMR-WB encoded speech is transmitted
   from a non-IP system (e.g., a GSM or 3GPP UMTS network) to an
   IP/UDP/RTP VoIP terminal, and/or vice versa, as depicted in Figure 3.













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

                     Figure 3: GW to VoIP terminal scenario

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

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

   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.










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   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/              |          IP network       |  GSM/
    3GPP UMTS network |                           |  3GPP UMTS network

                        Figure 4: GW to GW scenario

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

      -  the CMR value it receives on the IP side;

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

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

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

4.  AMR and AMR-WB RTP Payload Formats

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

4.1.  RTP Header Usage

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

   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.





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   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, in exact
   duplicates, in different AMR rate modes, or with data present in one
   packet and not present in another.  If multiple versions of the same
   speech frame are received, it is RECOMMENDED that the mode with the
   highest rate be used by the speech decoder.  A given frame MUST NOT
   be encoded as speech in one packet and comfort noise parameters in
   another.

   The payload length is always made an integral number of octets 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.







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4.2.  Payload Structure

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

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

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

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

4.3.  Bandwidth-Efficient Mode

4.3.1.  The Payload Header

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

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

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

   The codec mode request received in the CMR field is valid until the
   next codec mode request is received, i.e., a newly received CMR value
   corresponding to a speech mode, or NO_DATA overrides the previously
   received CMR value corresponding to a speech mode or NO_DATA.
   Therefore, if a terminal continuously wishes to receive frames in the





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   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 that is not a speech mode or
   NO_DATA, the CMR MUST be ignored by the receiver.

   In a multi-channel session, the codec mode request 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 codec mode request 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 codec 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 codec mode
   requests when sending speech to a multicast session but MAY use RTCP
   feedback information as a hint that a codec 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).  If the received CMR
   value is outside the signalled subset of modes, it MUST be ignored.


4.3.2.  The Payload Table of Contents

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






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

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

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

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

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

   NO_DATA (FT=15) frame could mean either that no data for that frame
   has been produced by the speech encoder 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 in any payload format configuration, except in the case
   of interleaving.  Also, frame-blocks containing only NO_DATA frames
   at the end of a packet SHOULD NOT be transmitted in any payload
   format configuration, except in the case of interleaving.  The AMR
   SCR/DTX is described in [6] and AMR-WB SCR/DTX in [7].

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

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




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   The frame quality indicator is included for interoperability with the
   ATM payload format described in ITU-T I.366.2, the UMTS Iu interface
   [20], 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
   more than 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 [12].  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 three 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 zero or more speech frames or
   comfort noise frames, as described in the ToC of the payload.

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





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

4.3.4.  Algorithm for Forming the Payload

   The complete RTP payload in bandwidth-efficient mode is formed by
   packing bits from the payload header, table of contents, and speech
   frames in order (as defined by their corresponding ToC entries in the
   ToC list), and to bring the payload to octet alignment, 0 to 7
   padding bits.  Padding bits MUST be set to zero and MUST be ignored
   on reception.  They are packed 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.




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   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 padding bits (P) 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|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

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 a 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
   kbps 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 are 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 zero bits are padded to the end to
   make the payload octet aligned.










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    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | CMR=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|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

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












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    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | CMR=15|1|1L FT=4|1|1|1R FT=4|1|1|2L FT=4|1|1|2R FT=4|1|1|3L FT|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |4|1|0|3R FT=4|1|d1L(0)                                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                               d1L(147)|d1R(0) |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   : ...                                                           :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                       d1R(147)|d2L(0)                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   : ...                                                           :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |d2L(147|d2R(0)                                                 |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   : ...                                                           :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                       d2R(147)|d3L(0)         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   : ...                                                           :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |               d3L(147)|d3R(0)                                 |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   : ...                                                           :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                                                       d3R(147)|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
















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4.4.  Octet-Aligned Mode

4.4.1.  The Payload Header

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

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

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

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

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

   ILP (4 bits, unsigned integer): This is an OPTIONAL field that is
      present only if interleaving is signalled.  ILP MUST take a value
      between 0 and ILL, inclusive, indicating the interleaving index
      for frame-blocks in this payload in the interleaving 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 interleaving group during an interleaving session.  Here
   we assume ILL=L for the interleaving group that starts at speech
   frame-block n.  We also assume that the first payload packet of the
   interleaving group is s, and the number of speech frame-blocks
   carried in each payload is N.  Then we will have:









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   Payload s (the first packet of this interleaving 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 interleaving 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 interleaving 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 interleaving group will start at frame-block n+N*(L+1).

   There will be no interleaving effect unless the number of frame-
   blocks per packet (N) is at least 2.  Moreover, the number of frame-
   blocks per payload (N) and the value of ILL MUST NOT be changed
   inside an interleaving group.  In other words, all payloads in an
   interleaving 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 media type 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
   interleaving group is less or equal to I, that is, 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.  That
   is, the ToC is as follows:

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




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   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 consecutively 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 three 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 results in the interleaving group size of 9 frame-blocks.


   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





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   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, and MUST be ignored on
           reception.

   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 bits long and corresponds to a speech frame (NOT a frame-
   block) carried in the payload.  Calculation and use of the CRC is
   specified in the next section.

4.4.2.1.  Use of Frame CRC for UED over IP

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

   To achieve UED, one SHOULD use a transport layer checksum (for
   example, the one defined in UDP-Lite [19]) to protect the IP,
   transport protocol (e.g., UDP-Lite), and RTP headers, as well as the
   payload header and the table of contents in the payload.  The frame
   CRC, when used, MUST be calculated only over all class A bits in the
   AMR or AMR-WB frame.  Class B and C bits in the AMR or AMR-WB 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.






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   If the transport layer checksum or link layer checksum detects any
   errors within the protected (sensitive) part, it is assumed that the
   complete packet will be discarded as defined by UDP-Lite [19].

   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 CRCs takes 8 bits

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

   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 appears as follows: 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 (LSB) bit of the CRC register and the bit.  The CRC



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   register is then right-shifted one step (each bit's significance is
   reduced by one), inputting a "0" as the leftmost bit (MSB).  If the
   result of the XOR operation mentioned above is a "1", then "10111000"
   is 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) has been used in this CRC calculation, the contents in
   CRC register should simply be copied to the corresponding field in
   the list of CRCs.

   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 zero
         bits at the end if all bits in the octet are not used.  The
         padding bits MUST be ignored on reception.  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 are arranged either
         one whole frame after another as usual, or with the octets of
         all frames interleaved together at the octet level, depending
         on the media type parameters negotiated for the payload type.
         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 [19]) or UEP
         (such as the ULP [22]) 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 payload type MUST be agreed via
   out-of-band means.  Section 8 specifies a media type 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 payload
   types.



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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
   octets 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 it is skipped in the robust sorting cycle.

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



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4.4.5.  Payload Examples

4.4.5.1.  Basic Single-Channel Payload Carrying Multiple Frames

   The following diagram shows an octet aligned payload from a single
   channel payload type 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 the above example, the last octet in both speech frames is
   padded with one zero bit 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
   payload type.  Two frame-blocks, each containing two 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, robust sorting, and frame-block interleaving are
   all enabled for the payload type.  The interleaving length is 2
   (ILL=1), and this payload is the first one in an interleaving group
   (ILP=0).





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   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 the above example, the last octet in all 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 media type
   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 operating 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 modes for a single audio





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   channel.  The other operating modes: interleaving, robust sorting,
   and frame-wise CRC (in both single and multi-channel) are OPTIONAL to
   implement.

   The mode-change-period, mode-change-capability, and mode-change-
   neighbor parameters are intended for signaling with GSM endpoints.
   When interoperability with GSM is desired, encoders SHOULD only
   perform codec mode changes to neighboring modes and in integer
   multiples of 40 ms (two frame-blocks), but decoders SHOULD accept
   codec mode changes at any time, i.e., for every frame-block.  The
   encoder may arbitrarily select the initial phase (odd or even frame-
   block) where codec mode changes are performed, but then SHOULD stick
   to that phase as far as possible.  However, in rare cases, handovers
   or other events (e.g., call forwarding) may change this phase and may
   also cause mode changes to non-neighboring modes.  The decoder SHALL
   therefore be prepared to accept changes also in the other phase and
   to other modes.  Section 8 specifies the usage of the parameters
   mode-change-period and mode-change-capability to indicate the desired
   behavior in applications.

   See 3GPP TS 26.103 [28] for preferred AMR and AMR-WB configurations
   for operation in GSM and 3GPP UMTS networks.  In gateway scenarios,
   encoders can be requested through the "mode-set" parameter to use a
   limited mode-set that is supported by the link beyond the gateway.
   Further, to avoid congestion on that link, the encoder SHOULD limit
   the initial codec mode for a session to a lower mode, until at least
   one frame-block is received with rate control information.

4.5.1.  Decoding Validation

   When processing a received payload packet, if the receiver finds that
   the calculated payload length, based on the information for the
   payload type 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.















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5.  AMR and AMR-WB Storage Format

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

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

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

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

   There also exists another storage format for AMR and AMR-WB that is
   suitable for applications with more advanced demands on the storage
   format, like random access or synchronization with video.  This
   format is the 3GPP-specified ISO-based multimedia file format 3GP
   [31].  Its media type is specified by RFC 3839 [32].

5.1.  Single-Channel Header

   A single-channel AMR or AMR-WB file header contains only a magic
   number.  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).






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   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.

   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 bits, unsigned integer): Indicates the number of audio
   channels contained in this storage file.  The valid values and the
   order of the channels within a frame-block are specified in Section
   4.1 in [12].




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5.3.  Speech Frames

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

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

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

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

   Following this one octet header come the speech bits as defined in
   4.4.3.  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 kbps 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|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   Non-received speech frames or 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]).  Frames or frame-blocks lost in
   transmission MUST be stored as NO_DATA frames or SPEECH_LOST (frame
   type 14, only available for AMR-WB) in complete frame-blocks to keep
   synchronization with the original media.

   Comfort noise frames of other types than AMR SID (FT=8) (i.e., frame
   type 9, 10, and 11 for AMR) SHALL NOT be used in the AMR file format.




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RFC 4867         RTP Payload Format for AMR and AMR-WB        April 2007


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, possibly "TCP Friendly Rate Control" [21].

7.  Security Considerations

   RTP packets using the payload format defined in this specification
   are subject to the general security considerations discussed in [8]
   and in any used profile, like AVP [12] or SAVP [26].

   As this format transports encoded speech, the main security issues
   include confidentiality, authentication, and integrity of the speech
   itself.  The payload format itself does not have any built-in
   security mechanisms.  External mechanisms, such as SRTP [26], need to
   be used for this functionality.  Note that the appropriate mechanism
   to provide security to RTP and the payloads following this memo may
   vary.  It is dependent on the application, the transport, and the
   signaling protocol employed.  Therefore, a single mechanism is not
   sufficient, although if suitable the usage of SRTP [26] is
   RECOMMENDED.  Other known mechanisms that may be used are IPsec [33]
   and TLS [34] (RTP over TCP), but other alternatives may also exist.

   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.





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RFC 4867         RTP Payload Format for AMR and AMR-WB        April 2007


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 of a need
   to encrypt the payload header or the table of contents due to a) that
   they only carry information about the requested speech mode, frame
   type, and frame quality, and b) that this information could be useful
   to some third party, e.g., quality monitoring.

   The packetization and unpacketization of the AMR and AMR-WB payload
   is done only at the endpoints.  Therefore encryption should be
   performed after packet encapsulation, and decryption should be
   performed before packet decapsulation.

   Encryption may affect interleaving.  Specifically, a change of keys
   should occur at the boundary between interleaving groups.  If it is
   not done at that boundary on both endpoints, the speech quality will
   be degraded during the complete interleaving group for any receiver.

   The encryption mechanism may impact the robustness of the error
   correcting mechanism.  This is discussed in Section 9.5 of SRTP [26].
   From this, UED/UEP based on robust sorting may be difficult to apply
   when the payload data is encrypted.


7.2.  Authentication and Integrity

   To authenticate the sender and to protect the integrity of the RTP
   packets in transit, an external mechanism has to be used.  As stated
   before, it is RECOMMENDED that SRTP [26] be used for common
   interoperability.  Note that the use of UED/UEP may be difficult to
   combine with some integrity protection mechanisms because any bit
   errors will cause the integrity check to fail.

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

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 media type registrations
   for the AMR and AMR-WB speech codecs.  The registrations are done
   following RFC 4855 [15] and the media registration rules [14].





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   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 media types or SDP.

   Two separate media type registrations are made, one for AMR and one
   for AMR-WB, because they are distinct encodings that must be
   distinguished by their own media type.

   Data formats are specified for both real-time transport in RTP and
   for storage type applications such as email attachments.

8.1.  AMR Media Type Registration

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

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

   Media Type name:     audio

   Media subtype name:  AMR

   Required parameters: none

   Optional parameters:

      These parameters apply to RTP transfer only.

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

      mode-set: Restricts the active codec mode set to a subset of all
               modes, for example, to be able to support transport
               channels such as GSM networks in gateway use cases.
               Possible values are a comma separated list of modes from
               the set: 0,...,7 (see Table 1a [2]).  The SID frame type
               8 and NO_DATA (frame type 15) are never included in the
               mode set, but can always be used.  If mode-set is
               specified, it MUST be abided, and frames encoded with
               modes outside of the subset MUST NOT be sent in any RTP
               payload or used in codec mode requests.  If not present,
               all codec modes are allowed for the payload type.




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      mode-change-period: Specifies a number of frame-blocks, N (1 or
               2), that is the frame-block period at which codec mode
               changes are allowed for the sender.  The initial phase of
               the interval is arbitrary, but changes must be separated
               by a period of N frame-blocks, i.e., a value of 2
               allows the sender to change mode every second frame-
               block.  The value of N SHALL be either 1 or 2.  If this
               parameter is not present, mode changes are allowed at
               any time during the session, i.e., N=1.

      mode-change-capability: Specifies if the client is capable to
               transmit with a restricted mode change period.  The
               parameter may take value of 1 or 2.  A value of 1
               indicates that the client is not capable of restricting
               the mode change period to 2, and that the codec mode may
               be changed at any point.  A value of 2 indicates that the
               client has the capability to restrict the mode change
               period to 2, and thus that the client can correctly
               interoperate with a receiver requiring a mode-change-
               period=2.  If this parameter is not present, the mode-
               change restriction capability is not supported, i.e.
               mode-change-capability=1.  To be able to interoperate
               fully with gateways to circuit switched networks (for
               example, GSM networks), transmissions with restricted
               mode changes (mode-change-capability=2) are required.
               Thus, clients RECOMMENDED to have the capability to
               support transmission according to
               mode-change-capability=2.

      mode-change-neighbor: Permissible values are 0 and 1.  If 1, the
               sender SHOULD only perform mode changes 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 that the
               media present in the packet represents.  The time SHOULD
               be an integer multiple of the frame size.  If this
               parameter is not present, the sender MAY encapsulate any
               number of speech frames into one RTP packet.






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

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

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

      ptime: see RFC 4566 [11].

      channels: The number of audio channels.  The possible values
               (1-6) and their respective channel order is specified in
               Section 4.1 in [12].  If omitted, it has the default
               value of 1.

      max-red: The maximum duration in milliseconds that elapses between
               the primary (first) transmission of a frame and any
               redundant transmission that the sender will use.  This
               parameter allows a receiver to have a bounded delay when
               redundancy is used.  Allowed values are between 0 (no
               redundancy will be used) and 65535.  If the parameter is
               omitted, no limitation on the use of redundancy is
               present.

   Encoding considerations:
        The Audio data is binary data, and must be encoded for non-
        binary transport; the Base64 encoding is suitable for email.
        When used in RTP context the data is framed as defined in [14].

   Security considerations:
        See Section 7 of RFC 4867.

   Public specification:
        RFC 4867
        3GPP TS 26.090, 26.092, 26.093, 26.101



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   Applications that use this media type:
        This media type is used in numerous applications needing
        transport or storage of encoded voice.  Some examples include;
        Voice over IP, streaming media, voice messaging, and voice
        recording on digital cameras.

   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: "amr " (fourth character is space)

        AMR speech frames may also be stored in the file format "3GP"
        defined in 3GPP TS 26.244 [31], which is identified using the
        media types "audio/3GPP" or "video/3GPP" as registered by RFC
        3839 [32].

   Person & email address to contact for further information:
        Magnus Westerlund <magnus.westerlund@ericsson.com>
        Ari Lakaniemi <ari.lakaniemi@nokia.com>

   Intended usage: COMMON.
        This media type is widely used in streaming, VoIP, and messaging
        applications on many types of devices.

   Restrictions on usage:
        When this media type is used in the context of transfer over
        RTP, the RTP payload format specified in Section 4 SHALL be
        used.  In all other contexts, the file format defined in Section
        5 SHALL be used.

   Author:
        Magnus Westerlund <magnus.westerlund@ericsson.com>
        Ari Lakaniemi <ari.lakaniemi@nokia.com>

   Change controller:
        IETF Audio/Video Transport working group delegated from the
        IESG.






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RFC 4867         RTP Payload Format for AMR and AMR-WB        April 2007


8.2.  AMR-WB Media Type Registration

   The media type for the Adaptive Multi-Rate Wideband (AMR-WB) codec is
   allocated from the IETF tree since AMR-WB is a widely used speech
   codec in general VoIP and messaging applications.  This media type
   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:  Restricts the active codec mode set to a subset of all
               modes, for example, to be able to support transport
               channels such as GSM networks in gateway use cases.
               Possible values are a comma-separated list of modes from
               the set: 0,...,8 (see Table 1a [4]).  The SID frame type
               9, SPEECH_LOST (frame type 14), and NO_DATA (frame type
               15) are never included in the mode set, but can always
               be used.  If mode-set is specified, it MUST be abided,
               and frames encoded with modes outside of the subset MUST
               NOT be sent in any RTP payload or used in codec mode
               requests.  If not present, all codec modes are allowed
               for the payload type.

      mode-change-period: Specifies a number of frame-blocks, N (1 or
               2), that is the frame-block period at which codec mode
               changes are allowed for the sender.  The initial phase of
               the interval is arbitrary, but changes must be separated
               by multiples of N frame-blocks, i.e., a value of 2
               allows the sender to change mode every second frame-
               block.  The value of N SHALL be either 1 or 2.  If this
               parameter is not present, mode changes are allowed at
               Any time during the session, i.e., N=1.





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      mode-change-capability: Specifies if the client is capable to
               transmit with a restricted mode change period.  The
               parameter may take value of 1 or 2.  A value of 1
               indicates that the client is not capable of restricting
               the mode change period to 2, and that the codec mode may
               be changed at any point.  A value of 2 indicates that the
               client has the capability to restrict the mode change
               period to 2, and thus that the client can correctly
               interoperate with a receiver requiring a mode-change-
               period=2.  If this parameter is not present, the mode-
               change restriction capability is not supported, i.e.
               mode-change-capability=1.  To be able to interoperate
               fully with gateways to circuit switched networks (for
               example, GSM networks), transmissions with restricted
               mode changes (mode-change-capability=2) are required.
               Thus, clients are RECOMMENDED to have the capability to
               support transmission according to
               mode-change-capability=2.

      mode-change-neighbor: Permissible values are 0 and 1.  If 1, the
               sender SHOULD only perform mode changes 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 that the
               media present in the packet represents.  The time SHOULD
               be an integer 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.  If 0 or not present, CRCs
               SHALL NOT be used.  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.





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      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 RFC 2327 [11].

      channels: The number of audio channels.  The possible values
               (1-6) and their respective channel order is specified in
               Section 4.1 in [12].  If omitted, it has the default
               value of 1.

      max-red: The maximum duration in milliseconds that elapses between
               the primary (first) transmission of a frame and any
               redundant transmission that the sender will use.  This
               parameter allows a receiver to have a bounded delay when
               redundancy is used.  Allowed values are between 0 (no
               redundancy will be used) and 65535.  If the parameter is
               omitted, no limitation on the use of redundancy is
               present.

   Encoding considerations:
        The Audio data is binary data, and must be encoded for non-
        binary transport; the Base64 encoding is suitable for email.
        When used in RTP context the data is framed as defined in [14].

   Security considerations:
        See Section 7 of RFC 4867.

   Public specification:
        RFC 4867
        3GPP TS 26.190, 26.192, 26.193, 26.201

   Applications that use this media type:
        This media type is used in numerous applications needing
        transport or storage of encoded voice.  Some examples include;
        Voice over IP, streaming media, voice messaging, and voice
        recording on digital cameras.









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RFC 4867         RTP Payload Format for AMR and AMR-WB        April 2007


   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: amrw
        Object identifier or OID: none

        AMR-WB speech frames may also be stored in the file format "3GP"
        defined in 3GPP TS 26.244 [31] and identified using the media
        type "audio/3GPP" or "video/3GPP" as registered by RFC 3839
        [32].

   Person & email address to contact for further information:
        Magnus Westerlund <magnus.westerlund@ericsson.com>
        Ari Lakaniemi <ari.lakaniemi@nokia.com>

   Intended usage: COMMON.
        This media type is widely used in streaming, VoIP, and messaging
        applications on many types of devices.

   Restrictions on usage:
        When this media type is used in the context of transfer over
        RTP, the RTP payload format specified in Section 4 SHALL be
        used.  In all other contexts, the file format defined in Section
        5 SHALL be used.

   Author:
        Magnus Westerlund <magnus.westerlund@ericsson.com>
        Ari Lakaniemi <ari.lakaniemi@nokia.com>

   Change controller:
        IETF Audio/Video Transport working group delegated from the
        IESG.

8.3.  Mapping Media Type Parameters into SDP

   The information carried in the 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:



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

      -  The media 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 are specified in Section 4.1 in [12].

      -  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 media type parameter string as a
         semicolon-separated list of parameter=value pairs.

8.3.1.  Offer-Answer Model Considerations

   The following considerations apply when using SDP Offer-Answer
   procedures to negotiate the use of AMR or AMR-WB payload in RTP:

      -  Each combination of the RTP payload transport format
         configuration parameters (octet-align, crc, robust-sorting,
         interleaving, and channels) is unique in its bit-pattern and
         not compatible with any other combination.  When creating an
         offer in an application desiring to use the more advanced
         features (crc, robust-sorting, interleaving, or more than one
         channel), the offerer is RECOMMENDED to also offer a payload
         type containing only the octet-aligned or bandwidth-efficient
         configuration with a single channel.  If multiple
         configurations are of interest to the application, they may all
         be offered; however, care should be taken not to offer too many
         payload types.  An SDP answerer MUST include, in the SDP answer
         for a payload type, the following parameters unmodified from
         the SDP offer (unless it removes the payload type): "octet-
         align"; "crc"; "robust-sorting"; "interleaving"; and
         "channels".  The SDP offerer and answerer MUST generate AMR or
         AMR-WB packets as described by these parameters.

      -  The "mode-set" parameter can be used to restrict the set of
         active AMR/AMR-WB modes used in a session.  This functionality
         is primarily intended for gateways to access networks such as
         GSM or 3GPP UMTS, where the access network may be capable of
         supporting only a subset of AMR/AMR-WB modes.  The 3GPP
         preferred codec configurations are defined in 3GPP TS 26.103
         [25], and it is RECOMMENDED that other networks also needing to
         restrict the mode set follow the preferred codec configurations
         defined in 3GPP for greatest interoperability.



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         The parameter is bi-directional, i.e., the restricted set
         applies to media both to be received and sent by the declaring
         entity.  If a mode set was supplied in the offer, the answerer
         SHALL return the mode-set unmodified or reject the payload
         type.  However, the answerer is free to choose a mode-set in
         the answer only if no mode-set was supplied in the offer for a
         unicast two-peer session.  The mode-set in the answer is
         binding both for offerer and answerer.  Thus, an offerer
         supporting all modes and subsets SHOULD NOT include the mode-
         set parameter.  For any other offerer it is RECOMMENDED to
         include each mode-set it can support as a separate payload type
         within the offer.  For multicast sessions, the answerer SHALL
         only participate in the session if it supports the offered
         mode-set.  Thus, it is RECOMMENDED that any offer for a
         multicast session include only the mode-set it will require the
         answerers to support, and that the mode-set be likely to be
         supported by all participants.

      -  The parameters "mode-change-period" and "mode-change-
         capability" are intended to be used in sessions with gateways,
         for example, when interoperating with GSM networks.  Both
         parameters are declarative and are combined to allow a session
         participant to determine if the payload type can be supported.
         The mode-change-period will indicate what the offerer or
         answerer requires of data it receives, while the mode-change-
         capability indicates its transmission capabilities.

         A mode-change-period=2 in the offer indicates a requirement on
         the answerer to send with a mode-change period of 2, i.e.,
         support mode-change-capability=2.  If the answerer requires
         mode-change-period=2, it SHALL only include it in the answer if
         the offerer either has indicated support with mode-change-
         capability=2 or has indicated mode-change-period=2; otherwise,
         the payload type SHALL be rejected.  An offerer that supports
         mode-change-capability=2 SHALL include the parameter in all
         offers to ensure the greatest possible interoperability, unless
         it includes mode-change-period=2 in the offer.  The mode-
         change-capability SHOULD be included in answers.  It is then
         indicating the answerer's capability to transmit with that
         mode-change-period for the provided payload format
         configuration.  The information is useful in future
         re-negotiation of the payload formats.

      -  The parameter "mode-change-neighbor" is a recommendation to
         restrict the switching of codec modes to its neighbor and
         SHOULD be followed.  It is intended to be used in gateway
         scenarios (for example, to GSM networks) where the support of




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         this parameter and the operations it implies improves
         interoperability.

         "mode-change-neighbor" is a declarative parameter.  By
         including the parameter, the offerer or answerer indicates that
         it desires to receive streams with "mode-change-neighbor"
         restrictions.

      -  In most cases, the parameters "maxptime" and "ptime" will not
         affect interoperability; however, the setting of the parameters
         can affect the performance of the application.  The SDP offer-
         answer handling of the "ptime" parameter is described in RFC
         3264 [13].  The "maxptime" parameter MUST be handled in the
         same way.

      -  The parameter "max-red" is a stream property parameter.  For
         send-only or send-recv unicast media streams, the parameter
         declares the limitation on redundancy that the stream sender
         will use.  For recvonly streams, it indicates the desired value
         for the stream sent to the receiver.  The answerer MAY change
         the value, but is RECOMMENDED to use the same limitation as the
         offer declares.  In the case of multicast, the offerer MAY
         declare a limitation; this SHALL be answered using the same
         value.  A media sender using this payload format is RECOMMENDED
         to always include the "max-red" parameter.  This information is
         likely to simplify the media stream handling in the receiver.
         This is especially true if no redundancy will be used, in which
         case "max-red" is set to 0.  As this parameter was not defined
         originally, some senders will not declare this parameter even
         if it will limit or not send redundancy at all.

      -  Any unknown parameter in an offer SHALL be removed in the
         answer.

8.3.2.  Usage of Declarative SDP

   In declarative usage, like SDP in RTSP [29] or SAP [30], the
   parameters SHALL be interpreted as follows:

   -  The payload format configuration parameters (octet-align, crc,
      robust-sorting, interleaving, and channels) are all declarative,
      and a participant MUST use the configuration(s) that is provided
      for the session.  More than one configuration may be provided if
      necessary by declaring multiple RTP payload types; however, the
      number of types should be kept small.






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   -  Any restriction of the AMR or AMR-WB encoder mode-switching and
      mode usage through the "mode-set", and "mode-change-period" MUST
      be followed by all participants of the session.  The restriction
      indicated by "mode-change-neighbor" SHOULD be followed.  Please
      note that such restrictions may be necessary if gateways to other
      transport systems like GSM participate in the session.  Failure to
      consider such restrictions may result in failure for a peer behind
      such a gateway to correctly receive all or parts of the session.
      Also, if different restrictions are needed by different peers in
      the same session (unless a common subset of the restrictions
      exists), some peer will not be able to participate.  Note that the
      usage of mode-change-capability is meaningless when no negotiation
      exists, and can thus be excluded in any declarations.

   -  Any "maxptime" and "ptime" values should be selected with care to
      ensure that the session's participants can achieve reasonable
      performance.

   -  The usage of "max-red" puts a global upper limit on the usage of
      redundancy that needs to be followed by all that understand the
      parameter.  However, due to the late addition of this parameter,
      it may be ignored by some implementations.

8.3.3.  Examples

   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.

   In an example of the usage of AMR in a possible GSM gateway-to-
   gateway scenario, the offerer is capable of supporting three
   different mode-sets and needs the mode-change-period to be 2 in
   combination with mode-change-neighbor restrictions.  The other
   gateway can only support two of these mode-sets and removes the
   payload type 97 in the answer.  If the offering GSM gateway only
   supports a single mode-set active at the same time, it should
   consider doing the 1 out of N selection procedures described in
   Section 10.2 of [13]:











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   Offer:

    m=audio 49120 RTP/AVP 97 98 99
    a=rtpmap:97 AMR/8000/1
    a=fmtp:97 mode-set=0,2,5,7; mode-change-period=2; \
      mode-change-capability=2; mode-change-neighbor=1
    a=rtpmap:98 AMR/8000/1
    a=fmtp:98 mode-set=0,2,3,6; mode-change-period=2; \
      mode-change-capability=2; mode-change-neighbor=1
    a=rtpmap:99 AMR/8000/1
    a=fmtp:99 mode-set=0,2,3,4; mode-change-period=2; \
      mode-change-capability=2; mode-change-neighbor=1
    a=maxptime:20

   Answer:

    m=audio 49120 RTP/AVP 98 99
    a=rtpmap:98 AMR/8000/1
    a=fmtp:98 mode-set=0,2,3,6; mode-change-period=2; \<
      mode-change-capability=2; mode-change-neighbor=1
    a=rtpmap:99 AMR/8000/1
    a=fmtp:99 mode-set=0,2,3,4; mode-change-period=2; \
      mode-change-capability=2; mode-change-neighbor=1
    a=maxptime:20

   The following example shows the usage of AMR between a non-GSM
   endpoint and a GSM gateway.  The non-GSM offerer requires no
   restrictions of the mode-change-period or mode-change-neighbor, but
   must signal its mode-change-capability in the offer and abide by
   those restrictions in the answer.

   Offer:

    m=audio 49120 RTP/AVP 97
    a=rtpmap:97 AMR/8000/1
    a=fmtp:97 mode-change-capability=2
    a=maxptime:20

   Answer:

    m=audio 49120 RTP/AVP 97
    a=rtpmap:97 AMR/8000/1
    a=fmtp:97 mode-set=0,2,4,7; mode-change-period=2; \
      mode-change-capability=2; mode-change-neighbor=1
    a=maxptime:20






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   Example of usage of AMR-WB in a possible VoIP scenario where UEP may
   be used (99) and a fallback declaration (98):

    m=audio 49120 RTP/AVP 99 98
    a=rtpmap:98 AMR-WB/16000
    a=fmtp:98 octet-align=1; mode-change-capability=2
    a=rtpmap:99 AMR-WB/16000
    a=fmtp:99 octet-align=1; crc=1; mode-change-capability=2

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

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

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

9.  IANA Considerations

   Two media types (audio/AMR and audio/AMR-WB) have been updated; see
   Section 8.

10.  Changes from RFC 3267

   The differences between RFC 3267 and this document are as follows:

   -  Added clarification of behavior in regards to mode change period
      and mode-change neighbor that is expected from an IP client; see
      Section 4.5.

   -  Updated the maxptime for better clarification.  The sentence that
      previously read: "The time SHOULD be a multiple of the frame
      size." now says "The time SHOULD be an integer multiple of the
      frame size."  This should have no impact on interoperability.

   -  Updated the definition of the mode-set parameter for
      clarification.

   -  Restricted the values for mode-change-period to 1 or 2, which are
      the values used in circuit-switched AMR systems.





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   -  Added a new media type parameter Mode-Change-Capability that
      defaults to 1, which is the assumed behavior of any non-updated
      implementation.  This enables the offer-answer procedures to work.

   -  Changed mode-change-neighbor to indicate a recommended behavior
      rather than a required one.

   -  Added an Offer-Answer Section, see Section 8.3.1.  This will have
      implications on the interoperability to implementations that have
      guessed how to perform offer/answer negotiation of the payload
      parameters.

   -  Clarified and aligned the unequal detection usage with the
      published UDP-Lite specification in Sections 3.6.1 and 4.4.2.1.
      This included replacing a normative statement about packet
      handling with an informative paragraph with a reference to UDP-
      Lite.

   -  Clarified the bit order in the CRC calculation in Section 4.4.2.1.

   -  Corrected the reference in Section 5.3 for the Q and FT fields.

   -  Changed the padding bit definition in Sections 4.4.2 and 5.3 so
      that it is clear that they shall be ignored.

   -  Added a clarification that comfort noise frames with frame type 9,
      10, and 11 SHALL NOT be used in the AMR file format.

   -  Clarified in Section 4.3.2 that the rules about not sending
      NO_DATA frames do apply for all payload format configurations with
      the exception of the interleaved mode.

   -  The reference list has been updated to now published RFCs: RFC
      3448, RFC 3550, RFC 3551, RFC 3711, RFC 3828, and RFC 4566.  A
      reference to 3GPP TS 26.101 has also been added.

   -  Added notes in storage format section and media type registration
      that AMR and AMR-WB frames can also be stored in the 3GP file
      format.

   -  Added a media type parameter "max-red" that allows the sender to
      declare a bounded usage of redundancy.  This parameter allows a
      receiver to optimize its function as it will know if redundancy
      will be used or not.  If it is used, the maximum extra delay
      introduced by the sender (that is needed to be considered by the
      receiver to fully utilize the redundancy) will be known.  The
      addition of this parameter should have no negative effects on
      older implementations as they are mandated to ignore unknown



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      parameters per RFC 3267.  In addition, older implementations are
      required to operate as if the value of max-red is unknown and
      possibly infinite.

   -  Updated the media type registration to comply with the new
      registration rules.

   -  Moved section on decoding validation from Security Considerations
      to Implementation Considerations, where it makes more sense.

   -  Clarified the application of encryption, integrity protection, and
      authentication mechanism to the payload.

11.  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 RFC 3267 and this replacement.  The authors would also
   like to thank Richard Ejzak, Thomas Belling, and Gorry Fairhurst for
   their input on this replacement of RFC 3267.

12.  References

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

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




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

   [8]  Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
        "RTP: A Transport Protocol for Real-Time Applications", STD 64,
        RFC 3550, July 2003.

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

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

   [11] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
        Description Protocol", RFC 4566, July 2006.

   [12] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video
        Conferences with Minimal Control", STD 65, RFC 3551, July 2003.

   [13] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
        Session Description Protocol (SDP)", RFC 3264, June 2002.

   [14] Freed, N. and J. Klensin, "Media Type Specifications and
        Registration Procedures", BCP 13, RFC 4288, December 2005.

   [15] Casner, S., "Media Type Registration of RTP Payload Formats",
        RFC 4855, February 2007.

12.2.  Informative References

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

   [17] 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.

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

   [19] Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E., and G.
        Fairhurst, "The Lightweight User Datagram Protocol (UDP-Lite)",
        RFC 3828, July 2004.



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   [20] 3GPP TS 25.415 "UTRAN Iu Interface User Plane Protocols",
        version 4.2.0 (2001-09), 3rd Generation Partnership Project
        (3GPP).

   [21] Handley, M., Floyd, S., Padhye, J., and J. Widmer, "TCP Friendly
        Rate Control (TFRC): Protocol Specification", RFC 3448, January
        2003.

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

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

   [24] 3GPP TS 26.102, "AMR speech codec interface to Iu and Uu",
        version 4.0.0 (2001-03), 3rd Generation Partnership Project
        (3GPP).

   [25] 3GPP TS 26.202, "AMR Wideband speech codec; Interface to Iu and
        Uu", version 5.0.0 (2001-03), 3rd Generation Partnership Project
        (3GPP).

   [26] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
        Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC
        3711, March 2004.

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

   [28] 3GPP TS 26.103, "Speech codec list for GSM and UMTS", version
        5.5.0 (2004-09), 3rd Generation Partnership Project (3GPP).

   [29] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time Streaming
        Protocol (RTSP)", RFC 2326, April 1998.

   [30] Handley, M., Perkins, C., and E. Whelan, "Session Announcement
        Protocol", RFC 2974, October 2000.

   [31] 3GPP TS 26.244, "3GPP file format (3GP)", version 6.1.0 (2004-
        09), 3rd Generation Partnership Project (3GPP).

   [32] Castagno, R. and D. Singer, "MIME Type Registrations for 3rd
        Generation Partnership Project (3GPP) Multimedia files", RFC
        3839, July 2004.

   [33] Kent, S. and K. Seo, "Security Architecture for the Internet
        Protocol", RFC 4301, December 2005.



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   [34] Dierks, T. and E. Rescorla, "The Transport Layer Security (TLS)
        Protocol Version 1.1", RFC 4346, April 2006.

   ETSI documents are available from <http://www.etsi.org/>.
   3GPP documents are available from <http://www.3gpp.org/>.
   TIA documents are available from <http://www.tiaonline.org/>.

Authors' Addresses

   Johan Sjoberg
   Ericsson AB
   SE-164 80 Stockholm, SWEDEN

   Phone: +46 8 7190000
   EMail: Johan.Sjoberg@ericsson.com


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

   Phone: +46 8 7190000
   EMail: Magnus.Westerlund@ericsson.com


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

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


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

   Phone: +1-847-632-3028
   EMail: Qiaobing.Xie@motorola.com









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Full Copyright Statement

   Copyright (C) The IETF Trust (2007).

   This document is subject to the rights, licenses and restrictions
   contained in BCP 78, and except as set forth therein, the authors
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Acknowledgement

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







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