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rfc:rfc4352

Network Working Group J. Sjoberg Request for Comments: 4352 M. Westerlund Category: Standards Track Ericsson

                                                          A. Lakaniemi
                                                             S. Wenger
                                                                 Nokia
                                                          January 2006
                     RTP Payload Format for the
    Extended Adaptive Multi-Rate Wideband (AMR-WB+) Audio Codec

Status of This Memo

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

Copyright Notice

 Copyright (C) The Internet Society (2006).

Abstract

 This document specifies a Real-time Transport Protocol (RTP) payload
 format for Extended Adaptive Multi-Rate Wideband (AMR-WB+) encoded
 audio signals.  The AMR-WB+ codec is an audio extension of the AMR-WB
 speech codec.  It encompasses the AMR-WB frame types and a number of
 new frame types designed to support high-quality music and speech.  A
 media type registration for AMR-WB+ is included in this
 specification.

Sjoberg, et al. Standards Track [Page 1] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

Table of Contents

 1. Introduction ....................................................3
 2. Definitions .....................................................4
    2.1. Glossary ...................................................4
    2.2. Terminology ................................................4
 3. Background of AMR-WB+ and Design Principles .....................4
    3.1. The AMR-WB+ Audio Codec ....................................4
    3.2. Multi-rate Encoding and Rate Adaptation ....................8
    3.3. Voice Activity Detection and Discontinuous Transmission ....8
    3.4. Support for Multi-Channel Session ..........................8
    3.5. Unequal Bit-Error Detection and Protection .................9
    3.6. Robustness against Packet Loss .............................9
         3.6.1. Use of Forward Error Correction (FEC) ...............9
         3.6.2. Use of Frame Interleaving ..........................10
    3.7. AMR-WB+ Audio over IP Scenarios ...........................11
    3.8. Out-of-Band Signaling .....................................11
 4. RTP Payload Format for AMR-WB+ .................................12
    4.1. RTP Header Usage ..........................................13
    4.2. Payload Structure .........................................14
    4.3. Payload Definitions .......................................14
         4.3.1. Payload Header .....................................14
         4.3.2. The Payload Table of Contents ......................15
         4.3.3. Audio Data .........................................20
         4.3.4. Methods for Forming the Payload ....................21
         4.3.5. Payload Examples ...................................21
    4.4. Interleaving Considerations ...............................24
    4.5. Implementation Considerations .............................25
         4.5.1. ISF Recovery in Case of Packet Loss ................26
         4.5.2. Decoding Validation ................................28
 5. Congestion Control .............................................28
 6. Security Considerations ........................................28
    6.1. Confidentiality ...........................................29
    6.2. Authentication and Integrity ..............................29
 7. Payload Format Parameters ......................................29
    7.1. Media Type Registration ...................................30
    7.2. Mapping Media Type Parameters into SDP ....................32
         7.2.1. Offer-Answer Model Considerations ..................32
         7.2.2. Examples ...........................................34
 8. IANA Considerations ............................................34
 9. Contributors ...................................................34
 10. Acknowledgements ..............................................34
 11. References ....................................................35
    11.1. Normative References .....................................35
    11.2. Informative References ...................................35

Sjoberg, et al. Standards Track [Page 2] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

1. Introduction

 This document specifies the payload format for packetization of
 Extended Adaptive Multi-Rate Wideband (AMR-WB+) [1] encoded audio
 signals into the Real-time Transport Protocol (RTP) [3].  The payload
 format supports the transmission of mono or stereo audio, aggregating
 multiple frames per payload, and mechanisms enhancing the robustness
 of the packet stream against packet loss.
 The AMR-WB+ codec is an extension of the Adaptive Multi-Rate Wideband
 (AMR-WB) speech codec.  New features include extended audio bandwidth
 to enable high quality for non-speech signals (e.g., music), native
 support for stereophonic audio, and the option to operate on, and
 switch between, several internal sampling frequencies (ISFs).  The
 primary usage scenario for AMR-WB+ is the transport over IP.
 Therefore, interworking with other transport networks, as discussed
 for AMR-WB in [7], is not a major concern and hence not addressed in
 this memo.
 The expected key application for AMR-WB+ is streaming.  To make the
 packetization process on a streaming server as efficient as possible,
 an octet-aligned payload format is desirable.  Therefore, a
 bandwidth-efficient mode (as defined for AMR-WB in [7]) is not
 specified herein; the bandwidth savings of the bandwidth-efficient
 mode would be very small anyway, since all extension frame types are
 octet aligned.
 The stereo encoding capability of AMR-WB+ renders the support for
 multi-channel transport at RTP payload format level, as specified for
 AMR-WB [7], obsolete.  Therefore, this feature is not included in
 this memo.
 This specification does not include a definition of a file format for
 AMR-WB+.  Instead, it refers to the ISO-based 3GP file format [14],
 which supports AMR-WB+ and provides all functionality required.  The
 3GP format also supports storage of AMR, AMR-WB, and many other
 multi-media formats, thereby allowing synchronized playback.
 The rest of the document is organized as follows: Background
 information on the AMR-WB+ codec, and design principles, can be found
 in Section 3.  The payload format itself is specified in Section 4.
 Sections 5 and 6 discuss congestion control and security
 considerations, respectively.  In Section 7, a media type
 registration is provided.

Sjoberg, et al. Standards Track [Page 3] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

2. Definitions

2.1. Glossary

 3GPP    - Third Generation Partnership Project
 AMR     - Adaptive Multi-Rate (Codec)
 AMR-WB  - Adaptive Multi-Rate Wideband (Codec)
 AMR-WB+ - Extended Adaptive Multi-Rate Wideband (Codec)
 CN      - Comfort Noise
 DTX     - Discontinuous Transmission
 FEC     - Forward Error Correction
 FT      - Frame Type
 ISF     - Internal Sampling Frequency
 SCR     - Source-Controlled Rate Operation
 SID     - Silence Indicator (the frames containing only CN
           parameters)
 TFI     - Transport Frame Index
 TS      - Timestamp
 VAD     - Voice Activity Detection
 UED     - Unequal Error Detection
 UEP     - Unequal Error Protection

2.2. Terminology

 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 [2].

3. Background of AMR-WB+ and Design Principles

 The Extended Adaptive Multi-Rate Wideband (AMR-WB+) [1] audio codec
 is designed to compress speech and audio signals at low bit-rate and
 good quality.  The codec is specified by the Third Generation
 Partnership Project (3GPP).  The primary target applications are 1)
 the packet-switched streaming service (PSS) [13], 2) multimedia
 messaging service (MMS) [18], and 3) multimedia broadcast and
 multicast service (MBMS) [19].  However, due to its flexibility and
 robustness, AMR-WB+ is also well suited for streaming services in
 other highly varying transport environments, for example, the
 Internet.

3.1. The AMR-WB+ Audio Codec

 3GPP originally developed the AMR-WB+ audio codec for streaming and
 messaging services in Global System for Mobile communications (GSM)
 and third generation (3G) cellular systems.  The codec is designed as
 an audio extension of the AMR-WB speech codec.  The extension adds
 new functionality to the codec in order to provide high audio quality

Sjoberg, et al. Standards Track [Page 4] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

 for a wide range of signals including music.  Stereophonic operation
 has also been added.  A new, high-efficiency hybrid stereo coding
 algorithm enables stereo operation at bit-rates as low as 6.2 kbit/s.
 The AMR-WB+ codec includes the nine frame types specified for AMR-WB,
 extended by new bit-rates ranging from 5.2 to 48 kbit/s.  The AMR-WB
 frame types can employ only a 16000 Hz sampling frequency and operate
 only on monophonic signals.  The newly introduced extension frame
 types, however, can operate at a number of internal sampling
 frequencies (ISFs), both in mono and stereo.  Please see Table 24 in
 [1] for details.  The output sampling frequency of the decoder is
 limited to 8, 16, 24, 32, or 48 kHz.
 An overview of the AMR-WB+ encoding operations is provided as
 follows.  The encoder receives the audio sampled at, for example, 48
 kHz.  The encoding process starts with pre-processing and resampling
 to the user-selected ISF.  The encoding is performed on equally sized
 super-frames.  Each super-frame corresponds to 2048 samples per
 channel, at the ISF.  The codec carries out a number of encoding
 decisions for each super-frame, thereby choosing between different
 encoding algorithms and block lengths, so as to achieve a fidelity-
 optimized encoding adapted to the signal characteristics of the
 source.  The stereo encoding (if used) executes separately from the
 monophonic core encoding, thus enabling the selection of different
 combinations of core and stereo encoding rates.  The resulting
 encoded audio is produced in four transport frames of equal length.
 Each transport frame corresponds to 512 samples at the ISF and is
 individually usable by the decoder, provided that its position in the
 super-frame structure is known.
 The codec supports 13 different ISFs, ranging from 12.8 to 38.4 kHz,
 as described by Table 24 of [1].  The high number of ISFs allows a
 trade-off between the audio bandwidth and the target bit-rate.  As
 encoding is performed on 2048 samples at the ISF, the duration of a
 super-frame and the effective bit-rate of the frame type in use
 varies.
 The ISF of 25600 Hz has a super-frame duration of 80 ms.  This is the
 'nominal' value used to describe the encoding bit-rates henceforth.
 Assuming this normalization, the ISF selection results in bit-rate
 variations from 1/2 up to 3/2 of the nominal bit-rate.
 The encoding for the extension modes is performed as one monophonic
 core encoding and one stereo encoding.  The core encoding is executed
 by splitting the monophonic signal into a lower and a higher
 frequency band.  The lower band is encoded employing either algebraic
 code excited linear prediction (ACELP) or transform coded excitation
 (TCX).  This selection can be made once per transport frame, but must

Sjoberg, et al. Standards Track [Page 5] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

 obey certain limitations of legal combinations within the super-
 frame.  The higher band is encoded using a low-rate parametric
 bandwidth extension approach.
 The stereo signal is encoded employing a similar frequency band
 decomposition; however, here the signal is divided into three bands
 that are individually parameterized.
 The total bit-rate produced by the extension is the result of the
 combination of the encoder's core rate, stereo rate, and ISF.  The
 extension supports 8 different core encoding rates, producing bit-
 rates between 10.4 and 24.0 kbit/s; see Table 22 in [1].  There are
 16 stereo encoding rates generating bit-rates between 2.0 and 8.0
 kbit/s; see Table 23 in [1].  The frame type uniquely identifies the
 AMR-WB modes, 4 fixed extension rates (see below), 24 combinations of
 core and stereo rates for stereo signals, and the 8 core rates for
 mono signals, as listed in Table 25 in [1].  This implies that the
 AMR-WB+ supports encoding rates between 10.4 and 32 kbit/s, assuming
 an ISF of 25600 Hz.
 Different ISFs allow for additional freedom in the produced bit-rates
 and audio quality.  The selection of an ISF changes the available
 audio bandwidth of the reconstructed signal, and also the total bit-
 rate.  The bit-rate for a given combination of frame type and ISF is
 determined by multiplying the frame type's bit-rate with the used
 ISF's bit-rate factor; see Table 24 in [1].
 The extension also has four frame types which have fixed ISFs.
 Please see frame types 10-13 in Table 21 in [1].  These four pre-
 defined frame types have a fixed input sampling frequency at the
 encoder, which can be set at either 16 or 24 kHz.  Like the AMR-WB
 frame types, transport frames encoded utilizing these frame types
 represent exactly 20 ms of the audio signal.  However, they are also
 part of 80 ms super-frames.  Frame types 0-13 (AMR-WB and fixed
 extension rates), as listed in Table 21 in [1], do not require an
 explicit ISF indication.  The other frame types, 14-47, require the
 ISF employed to be indicated.
 The 32 different frame types of the extension, in combination with 13
 ISFs, allows for a great flexibility in bit-rate and selection of
 desired audio quality.  A number of combinations exist that produce
 the same codec bit-rate.  For example, a 32 kbit/s audio stream can
 be produced by utilizing frame type 41 (i.e., 25.6 kbit/s) and the
 ISF of 32kHz (5/4 * (19.2+6.4) = 32 kbit/s), or frame type 47 and the
 ISF of 25.6 kHz (1 * (24 + 8) = 32 kbit/s).  Which combination is
 more beneficial for the perceived audio quality depends on the
 content.  In the above example, the first case provides a higher
 audio bandwidth, while the second one spends the same number of bits

Sjoberg, et al. Standards Track [Page 6] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

 on somewhat narrower audio bandwidth but provides higher fidelity.
 Encoders are free to select the combination they deem most
 beneficial.
 Since a transport frame always corresponds to 512 samples at the used
 ISF, its duration is limited to the range 13.33 to 40 ms; see Table
 1.  An RTP Timestamp clock rate of 72000 Hz, as mandated by this
 specification, results in AMR-WB+ transport frame lengths of 960 to
 2880 timestamp ticks, depending solely on the selected ISF.
    Index   ISF   Duration(ms) Duration(TS Ticks @ 72 kHz)
    ------------------------------------------------------
      0     N/A      20             1440
      1    12800     40             2880
      2    14400     35.55          2560
      3    16000     32             2304
      4    17067     30             2160
      5    19200     26.67          1920
      6    21333     24             1728
      7    24000     21.33          1536
      8    25600     20             1440
      9    28800     17.78          1280
     10    32000     16             1152
     11    34133     15             1080
     12    36000     14.22          1024
     13    38400     13.33           960
    Table 1: Normative number of RTP Timestamp Ticks for each
             Transport Frame depending on ISF (ISF and Duration in
             ms are rounded)
 The encoder is free to change both the ISF and the encoding frame
 type (both mono and stereo) during a session.  For the extension
 frame types with index 10-13 and 16-47, the ISF and frame type
 changes are constrained to occur at super-frame boundaries.  This
 implies that, for the frame types mentioned, the ISF is constant
 throughout a super-frame.  This limitation does not apply for frame
 types with index 0-9, 14, and 15; i.e., the original AMR-WB frame
 types.
 A number of features of the AMR-WB+ codec require special
 consideration from a transport point of view, and solutions that
 could perhaps be viewed as unorthodox.  First, there are constraints
 on the RTP timestamping, due to the relationship of the frame
 duration and the ISFs.  Second, each frame of encoded audio must
 maintain information about its frame type, ISF, and position in the
 super-frame.

Sjoberg, et al. Standards Track [Page 7] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

3.2. Multi-rate Encoding and Rate Adaptation

 The multi-rate encoding capability of AMR-WB+ is designed to preserve
 high audio quality under a wide range of bandwidth requirements and
 transmission conditions.
 AMR-WB+ enables seamless switching between frame types that use the
 same number of audio channels and the same ISF.  Every AMR-WB+ codec
 implementation is required to support all frame types defined by the
 codec and must be able to handle switching between any two frame
 types.  Switching between frame types employing a different number of
 audio channels or a different ISF must also be supported, but it may
 not be completely seamless.  Therefore, it is recommended to perform
 such switching infrequently and, if possible, during periods of
 silence.

3.3. Voice Activity Detection and Discontinuous Transmission

 AMR-WB+ supports the same algorithms as AMR-WB for voice activity
 detection (VAD) and generation of comfort noise (CN) parameters
 during silence periods.  However, these functionalities can only be
 used in conjunction with the AMR-WB frame types (FT=0-8).  This
 option allows reducing 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-WB+ frames containing CN parameters are
 called Silence Indicator (SID) frames.  More details about the VAD
 and DTX functionality are provided in [4] and [5].

3.4. Support for Multi-Channel Session

 Some of the AMR-WB+ frame types support the encoding of stereophonic
 audio.  Because of this native support for a two-channel stereophonic
 signal, it does not seem necessary to support multi-channel transport
 with separate codec instances, as specified in the AMR-WB RTP payload
 [7].  The codec has the capability of stereo to mono downmixing as
 part of the decoding process.  Thus, a receiver that is only capable
 of playout of monophonic audio must still be able to decode and play
 signals originally encoded and transmitted as stereo.  However, to
 avoid spending bits on a stereo encoding that is not going to be
 utilized, a mechanism is defined in this specification to signal
 mono-only audio.

Sjoberg, et al. Standards Track [Page 8] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

3.5. Unequal Bit-Error Detection and Protection

 The audio bits encoded in each AMR-WB frame are sorted according to
 their different perceptual sensitivity to bit errors.  In cellular
 systems, for example, this property can be exploited to achieve
 better voice quality, by using unequal error protection and detection
 (UEP and UED) mechanisms.  However, the bits of the extension frame
 types of the AMR-WB+ codec do not have a consistent perceptual
 significance property and are not sorted in this order.  Thus, UEP or
 UED is meaningless with the extension frame types.  If there is a
 need to use UEP or UED for AMR-WB frame types, it is recommended that
 RFC 3267 [7] be used.

3.6. Robustness against Packet Loss

 The payload format supports two mechanisms to improve robustness
 against packet loss: simple forward error correction (FEC) and frame
 interleaving.

3.6.1. Use of Forward Error Correction (FEC)

 Generic forward error correction within RTP is defined, for example,
 in RFC 2733 [11].  Audio redundancy coding is defined in RFC 2198
 [12].  Either scheme can be used to add redundant information to the
 RTP packet stream and make it more resilient to packet losses, at the
 expense of a higher bit rate.  Please see either RFC for a discussion
 of the implications of the higher bit rate to network congestion.
 In addition to these media-unaware mechanisms, this memo specifies an
 AMR-WB+ specific form of audio redundancy coding, which may be
 beneficial in terms of packetization overhead.
 Conceptually, previously transmitted transport frames are aggregated
 together with new ones.  A sliding window is used to group the frames
 to be sent in each payload.  Figure 1 below shows an example.
  1. -+——–+——–+——–+——–+——–+——–+——–+–

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

  1. -+——–+——–+——–+——–+——–+——–+——–+–
   <---- p(n-1) ---->
            <----- p(n) ----->
                     <---- p(n+1) ---->
                              <---- p(n+2) ---->
                                       <---- p(n+3) ---->
                                                <---- p(n+4) ---->
 Figure 1: An example of redundant transmission

Sjoberg, et al. Standards Track [Page 9] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

 Here, each frame is retransmitted once in the following RTP payload
 packet.  F(n-2)...f(n+4) denote a sequence of audio frames, and
 p(n-1)...p(n+4) a sequence of payload packets.
 The mechanism described does not require signaling at the session
 setup.  In other words, the audio sender can choose to use this
 scheme without consulting the receiver.  For a certain timestamp, the
 receiver may receive multiple copies of a frame containing encoded
 audio data or frames indicated as NO_DATA.  The cost of this scheme
 is bandwidth and the receiver delay necessary to allow the redundant
 copy to arrive.
 This redundancy scheme provides a functionality similar to the one
 described in RFC 2198, but it works only if both original frames and
 redundant representations are AMR-WB+ frames.  When the use of other
 media coding schemes is desirable, one has to resort to RFC 2198.
 The sender is responsible for selecting an appropriate amount of
 redundancy based on feedback about the channel conditions, e.g., in
 the RTP Control Protocol (RTCP) [3] receiver reports.  The sender is
 also responsible for avoiding congestion, which may be exacerbated by
 redundancy (see Section 5 for more details).

3.6.2. Use of Frame Interleaving

 To decrease protocol overhead, the payload design allows several
 audio transport frames to be encapsulated into a single RTP packet.
 One of the drawbacks of such an approach is that in case of packet
 loss several consecutive frames are lost.  Consecutive frame loss
 normally renders error concealment less efficient and usually causes
 clearly audible and annoying distortions in the reconstructed audio.
 Interleaving of transport frames can improve the audio quality in
 such cases by distributing the consecutive losses into a number of
 isolated frame losses, which are easier to conceal.  However,
 interleaving and bundling several frames per payload also increases
 end-to-end delay and sets higher buffering requirements.  Therefore,
 interleaving is not appropriate for all use cases or devices.
 Streaming applications should most likely be able to exploit
 interleaving to improve audio quality in lossy transmission
 conditions.
 Note that this payload design supports the use of frame interleaving
 as an option.  The usage of this feature needs to be negotiated in
 the session setup.
 The interleaving supported by this format is rather flexible.  For
 example, a continuous pattern can be defined, as depicted in Figure
 2.

Sjoberg, et al. Standards Track [Page 10] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

  1. -+——–+——–+——–+——–+——–+——–+——–+–

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

  1. -+——–+——–+——–+——–+——–+——–+——–+–
            [ P(n)   ]
   [ P(n+1) ]                 [ P(n+1) ]
                     [ P(n+2) ]                 [ P(n+2) ]
                                       [ P(n+3) ]                 [P(
                                                         [ P(n+4) ]
 Figure 2: An example of interleaving pattern that has constant delay
 In Figure 2 the consecutive frames, denoted f(n-2) to f(n+4), are
 aggregated into packets P(n) to P(n+4), each packet carrying two
 frames.  This approach provides an interleaving pattern that allows
 for constant delay in both the interleaving and deinterleaving
 processes.  The deinterleaving buffer needs to have room for at least
 three frames, including the one that is ready to be consumed.  The
 storage space for three frames is needed, for example, when f(n) is
 the next frame to be decoded: since frame f(n) was received in packet
 P(n+2), which also carried frame f(n+3), both these frames are stored
 in the buffer.  Furthermore, frame f(n+1) received in the previous
 packet, P(n+1), is also in the deinterleaving buffer.  Note also that
 in this example the buffer occupancy varies: when frame f(n+1) is the
 next one to be decoded, there are only two frames, f(n+1) and f(n+3),
 in the buffer.

3.7. AMR-WB+ Audio over IP Scenarios

 Since the primary target application for the AMR-WB+ codec is
 streaming over packet networks, the most relevant usage scenario for
 this payload format is IP end-to-end between a server and a terminal,
 as shown in Figure 3.
            +----------+                          +----------+
            |          |    IP/UDP/RTP/AMR-WB+    |          |
            |  SERVER  |<------------------------>| TERMINAL |
            |          |                          |          |
            +----------+                          +----------+
             Figure 3: Server to terminal IP scenario

3.8. Out-of-Band Signaling

 Some of the options of this payload format remain constant throughout
 a session.  Therefore, they can be controlled/negotiated at the
 session setup.  Throughout this specification, these options and
 variables are denoted as "parameters to be established through out-

Sjoberg, et al. Standards Track [Page 11] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

 of-band means".  In Section 7, all the parameters are formally
 specified in the form of media type registration for the AMR-WB+
 encoding.  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 7.2 provides a mapping of
 the parameters into the Session Description Protocol (SDP) [6] for
 those applications that use SDP.

4. RTP Payload Format for AMR-WB+

 The main emphasis in the payload design for AMR-WB+ has been to
 minimize the overhead in typical use cases, while providing full
 flexibility with a slightly higher overhead.  In order to keep the
 specification reasonably simple, we refrained from defining frame-
 specific parameters for each frame type.  Instead, a few common
 parameters were specified that cover all types of frames.
 The payload format has two modes: basic mode and interleaved mode.
 The main structural difference between the two modes is the extension
 of the table of content entries with frame displacement fields when
 operating in the interleaved mode.  The basic mode supports
 aggregation of multiple consecutive frames in a payload.  The
 interleaved mode supports aggregation of multiple frames that are
 non-consecutive in time.  In both modes it is possible to have frames
 encoded with different frame types in the same payload.  The ISF must
 remain constant throughout the payload of a single packet.
 The payload format is designed around the property of AMR-WB+ frames
 that the frames are consecutive in time and share the same frame
 duration (in the absence of an ISF change).  This enables the
 receiver to derive the timestamp for an individual frame within a
 payload.  In basic mode, the deriving process is based on the order
 of frames.  In interleaved mode, it is based on the compact
 displacement fields.  The frame timestamps are used to regenerate the
 correct order of frames after reception, identify duplicates, and
 detect lost frames that require concealment.
 The interleaving scheme of this payload format is significantly more
 flexible than the one specified in RFC 3267.  The AMR and AMR-WB
 payload format is only capable of using periodic patterns with frames
 taken from an interleaving group at fixed intervals.  The
 interleaving scheme of this specification, in contrast, allows for
 any interleaving pattern, as long as the distance in decoding order
 between any two adjacent frames is not more than 256 frames.  Note
 that even at the highest ISF this allows an interleaving depth of up
 to 3.41 seconds.

Sjoberg, et al. Standards Track [Page 12] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

 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 audio frame multiple times.  All
 redundantly sent frames MUST use the same frame type and ISF, and
 MUST have the same RTP timestamp, or MUST be a NO_DATA frame (FT=15).
 The payload consists of octet-aligned elements (header, ToC, and
 audio frames).  Only the audio frames for AMR-WB frame types (0-9)
 require padding for octet alignment.  If additional padding is
 desired, then the P bit in the RTP header MAY be set, and padding MAY
 be appended as specified in [3].

4.1. RTP Header Usage

 The format of the RTP header is specified in [3].  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 in the packet.  The timestamp
 clock frequency SHALL be 72000 Hz.  This frequency allows the frame
 duration to be integer RTP timestamp ticks for the ISFs specified in
 Table 1.  It also provides reasonable conversion factors to the
 input/output audio sampling frequencies supported by the codec.  See
 Section 4.3.2.3 for guidance on how to derive the RTP timestamp for
 any audio frame beyond the first one.
 The RTP header marker bit (M) SHALL be set to 1 whenever the first
 frame carried in the packet is the first frame in a talkspurt (see
 the definition of talkspurt in Section 4.1 of [9]).  For all other
 packets, the marker bit SHALL be set to zero (M=0).
 The assignment of an RTP payload type for the format defined in this
 memo is outside the scope of this document.  The RTP profile in use
 either assigns a static payload type or mandates binding the payload
 type dynamically.
 The media type parameter "channels" is used to indicate the maximum
 number of channels allowed for a given payload type.  A payload type
 where channels=1 (mono) SHALL only carry mono content.  A payload
 type for which channels=2 has been declared MAY carry both mono and
 stereo content.  Note that this definition is different from the one
 in RFC 3551 [9].  As mentioned before, the AMR-WB+ codec handles the
 support of stereo content and the (eventual) downmixing of stereo to
 mono internally.  This makes it unnecessary to negotiate for the
 number of channels for reasons other than bit-rate efficiency.

Sjoberg, et al. Standards Track [Page 13] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

4.2. Payload Structure

 The payload consists of a payload header, a table of contents, and
 the audio data representing one or more audio frames.  The following
 diagram shows the general payload format layout:
 +----------------+-------------------+----------------
 | payload header | table of contents | audio data ...
 +----------------+-------------------+----------------
 Payloads containing more than one audio frame are called compound
 payloads.
 The following sections describe the variations taken by the payload
 format depending on the mode in use: basic mode or interleaved mode.

4.3. Payload Definitions

4.3.1. Payload Header

 The payload header carries data that is common for all frames in the
 payload.  The structure of the payload header is described below.
  0 1 2 3 4 5 6 7
 +-+-+-+-+-+-+-+-+
 |   ISF   |TFI|L|
 +-+-+-+-+-+-+-+-+
 ISF (5 bits): Indicates the Internal Sampling Frequency employed for
    all frames in this payload.  The index value corresponds to
    internal sampling frequency as specified in Table 24 in [1].  This
    field SHALL be set to 0 for payloads containing frames with Frame
    Type values 0-13.
 TFI (2 bits): Transport Frame Index, from 0 (first) to 3 (last),
    indicating the position of the first transport frame of this
    payload in the AMR-WB+ super-frame structure.  For payloads with
    frames of only Frame Type values 0-9, this field SHALL be set to 0
    by the sender.  The TFI value for a frame of type 0-9 SHALL be
    ignored by the receiver.  Note that the frame type is coded in the
    table of contents (as discussed later); hence, the mentioned
    dependencies of the frame type can be applied easily by
    interpreting only values carried in the payload header.  It is not
    necessary to interpret the audio bit stream itself.

Sjoberg, et al. Standards Track [Page 14] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

 L (1 bit): Long displacement field flag for payloads in interleaved
    mode.  If set to 0, four-bit displacement fields are used to
    indicate interleaving offset; if set to 1, displacement fields of
    eight bits are used (see Section 4.3.2.2).  For payloads in the
    basic mode, this bit SHALL be set to 0 and SHALL be ignored by the
    receiver.
 Note that frames employing different ISF values require encapsulation
 in separate packets.  Thus, special considerations apply when
 generating interleaved packets and an ISF change is executed.  In
 particular, frames that, according to the previously used
 interleaving pattern, would be aggregated into a single packet have
 to be separated into different packets, so that the aforementioned
 condition (all frames in a packet share the ISF) remains true.  A
 naive implementation that splits the frames with different ISF into
 different packets can result in up to twice the number of RTP
 packets, when compared to an optimal interleaved solution.
 Alteration of the interleaving before and after the ISF change may
 reduce the need for extra RTP packets.

4.3.2. The Payload Table of Contents

 The table of contents (ToC) consists of a list of entries, each entry
 corresponds to a group of audio frames carried in the payload, as
 depicted below.
 +----------------+----------------+- ... -+----------------+
 |  ToC entry #1  |  ToC entry #2  |          ToC entry #N  |
 +----------------+----------------+- ... -+----------------+
 When multiple groups of frames are present in a payload, the ToC
 entries SHALL be placed in the packet in order of increasing RTP
 timestamp value (modulo 2^32) of the first transport frame the TOC
 entry represents.

4.3.2.1. ToC Entry in the Basic Mode

 A ToC entry of a payload in the basic mode has the following format:
  0                   1
  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |F| Frame Type  |    #frames    |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 F (1 bit): If set to 1, indicates that this ToC entry is followed by
    another ToC entry; if set to 0, indicates that this ToC entry is
    the last one in the ToC.

Sjoberg, et al. Standards Track [Page 15] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

 Frame Type (FT) (7 bits): Indicates the audio codec frame type used
    for the group of frames referenced by this ToC entry.  FT
    designates the combination of AMR-WB+ core and stereo rate, one of
    the special AMR-WB+ frame types, the AMR-WB rate, or comfort
    noise, as specified by Table 25 in [1].
 #frames (8 bits): Indicates the number of frames in the group
    referenced by this ToC entry.  ToC entries with this field equal
    to 0 (which would indicate zero frames) SHALL NOT be used, and
    received packets with such a TOC entry SHALL be discarded.

4.3.2.2. ToC Entry in the Interleaved Mode

 Two different ToC entry formats are defined in interleaved mode.
 They differ in the length of the displacement field, 4 bits or 8
 bits.  The L-bit in the payload header differentiates between the two
 modes.
 If L=0, a ToC entry has the following 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
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |F| Frame Type  |    #frames    |  DIS1 |  ...  |  DISi |  ...  |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |  ...  |  ...  |  DISn |  Padd |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 F (1 bit): See definition in 4.3.2.1.
 Frame Type (FT) (7 bits): See definition in 4.3.2.1.
 #frames (8 bits): See definition in 4.3.2.1.
 DIS1...DISn (4 bits): A list of n (n=#frames) displacement fields
    indicating the displacement of the i:th (i=1..n) audio frame
    relative to the preceding audio frame in the payload, in units of
    frames.  The four-bit unsigned integer displacement values may be
    between 0 and 15, indicating the number of audio frames in
    decoding order between the (i-1):th and the i:th frame in the
    payload.  Note that for the first ToC entry of the payload, the
    value of DIS1 is meaningless.  It SHALL be set to zero by a sender
    and SHALL be ignored by a receiver.  This frame's location in the
    decoding order is uniquely defined by the RTP timestamp and TFI in
    the payload header.  Note also that for subsequent ToC entries,
    DIS1 indicates the number of frames between the last frame of the
    previous group and the first frame of this group.

Sjoberg, et al. Standards Track [Page 16] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

 Padd (4 bits): To ensure octet alignment, four padding bits SHALL be
    included at the end of the ToC entry in case there is odd number
    of frames in the group referenced by this entry.  These bits SHALL
    be set to zero and SHALL be ignored by the receiver.  If a group
    containing an even number of frames is referenced by this ToC
    entry, these padding bits SHALL NOT be included in the payload.
 If L=1, a ToC entry has the following 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
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |F| Frame Type  |    #frames    |      DIS1     |      ...      |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |      ...      |     DISn      |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 F (1 bit): See definition in 4.3.2.1.
 Frame Type (FT) (7 bits): See definition in 4.3.2.1.
 #frames (8 bits): See definition in 4.3.2.1.
 DIS1...DISn (8 bits): A list of n (n=#frames) displacement fields
    indicating the displacement of the i:th (i=1..n) audio frame
    relative to the preceding audio frame in the payload, in units of
    frames.  The eight-bit unsigned integer displacement values may be
    between 0 and 255, indicating the number of audio frames in
    decoding order between the (i-1):th and the i:th frame in the
    payload.  Note that for the first ToC entry of the payload, the
    value of DIS1 is meaningless.  It SHALL be set to zero by a sender
    and SHALL be ignored by a receiver.  This frame's location in the
    decoding order is uniquely defined by the RTP timestamp and TFI in
    the payload header.  Note also that for subsequent ToC entries,
    DIS1 indicates the displacement between the last frame of the
    previous group and the first frame of this group.

4.3.2.3. RTP Timestamp Derivation

 The RTP Timestamp value for a frame SHALL be the timestamp value of
 the first audio sample encoded in the frame.  The timestamp value for
 a frame is derived differently depending on the payload mode, basic
 or interleaved.  In both cases, the first frame in a compound packet
 has an RTP timestamp equal to the one received in the RTP header.  In
 the basic mode, the RTP time for any subsequent frame is derived in
 two steps.  First, the sum of the frame durations (see Table 1) of
 all the preceding frames in the payload is calculated.  Then, this
 sum is added to the RTP header timestamp value.  For example, let's

Sjoberg, et al. Standards Track [Page 17] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

 assume that the RTP Header timestamp value is 12345, the payload
 carries four frames, and the frame duration is 16 ms (ISF = 32 kHz)
 corresponding to 1152 timestamp ticks.  Then the RTP timestamp of the
 fourth frame in the payload is 12345 + 3 * 1152 = 15801.
 In interleaved mode, the RTP timestamp for each frame in the payload
 is derived from the RTP header timestamp and the sum of the time
 offsets of all preceding frames in this payload.  The frame
 timestamps are computed based on displacement fields and the frame
 duration derived from the ISF value.  Note that the displacement in
 time between frame i-1 and frame i is (DISi + 1) * frame duration
 because the duration of the (i-1):th must also be taken into account.
 The timestamp of the first frame of the first group of frames (TS(1))
 (i.e., the first frame of the payload) is the RTP header timestamp.
 For subsequent frames in the group, the timestamp is computed by
    TS(i) = TS(i-1) + (DISi + 1) * frame duration,    2 < i < n
 For subsequent groups of frames, the timestamp of the first frame is
 computed by
    TS(1) = TSprev + (DIS1 + 1) * frame duration,
 where TSprev denotes the timestamp of the last frame in the previous
 group.  The timestamps of the subsequent frames in the group are
 computed in the same way as for the first group.
 The following example derives the RTP timestamps for the frames in an
 interleaved mode payload having the following header and ToC
 information:
 RTP header timestamp: 12345
 ISF = 32 kHz
 Frame 1 displacement field: DIS1 = 0
 Frame 2 displacement field: DIS2 = 6
 Frame 3 displacement field: DIS3 = 4
 Frame 4 displacement field: DIS4 = 7
 Assuming an ISF of 32 kHz, which implies a frame duration of 16 ms,
 one frame lasts 1152 ticks.  The timestamp of the first frame in the
 payload is the RTP timestamp, i.e., TS(1) = RTP TS.  Note that the
 displacement field value for this frame must be ignored.  For the
 second frame in the payload, the timestamp can be calculated as TS(2)
 = TS(1) + (DIS2 + 1) * 1152 = 20409.  For the third frame, the
 timestamp is TS(3) = TS(2) + (DIS3 + 1) * 1152 = 26169.  Finally, for
 the fourth frame of the payload, we have TS(4) = TS(3) + (DIS4 + 1) *
 1152 = 35385.

Sjoberg, et al. Standards Track [Page 18] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

4.3.2.4. Frame Type Considerations

 The value of Frame Type (FT) is defined in Table 25 in [1].  FT=14
 (AUDIO_LOST) is used to denote frames that are lost.  A NO_DATA
 (FT=15) frame could result from two situations: First, that no data
 has been produced by the audio encoder; and second, that no data is
 transmitted in the current payload.  An example for the latter would
 be that the frame in question has been or will be sent in an earlier
 or later packet.  The duration for these non-included frames is
 dependent on the internal sampling frequency indicated by the ISF
 field.
 For frame types with index 0-13, the ISF field SHALL be set 0.  The
 frame duration for these frame types is fixed to 20 ms in time, i.e.,
 1440 ticks in 72 kHz.  For payloads containing only frames of type
 0-9, the TFI field SHALL be set to 0 and SHALL be ignored by the
 receiver.  In a payload combining frames of type 0-9 and 10-13, the
 TFI values need to be set to match the transport frames of type
 10-13.  Thus, frames of type 0-9 will also have a derived TFI, which
 is ignored.

4.3.2.5. Other TOC Considerations

 If a ToC entry with an undefined FT value is received, the whole
 packet SHALL be discarded.  This is to avoid the loss of data
 synchronization in the depacketization process, which can result in a
 severe degradation in audio quality.
 Packets containing only NO_DATA frames SHOULD NOT be transmitted.
 Also, NO_DATA frames at the end of a frame sequence to be carried in
 a payload SHOULD NOT be included in the transmitted packet.  The
 AMR-WB+ SCR/DTX is identical with AMR-WB SCR/DTX described in [5] and
 can only be used in combination with the AMR-WB frame types (0-8).
 When multiple groups of frames are present, their ToC entries SHALL
 be placed in the ToC in order of increasing RTP timestamp value
 (modulo 2^32) of the first transport frame the TOC entry represents,
 independent of the payload mode.  In basic mode, the frames SHALL be
 consecutive in time, while in interleaved mode the frames MAY not
 only be non-consecutive in time but MAY even have varying inter-frame
 distances.

4.3.2.6. ToC Examples

 The following example illustrates a ToC for three audio frames in
 basic mode.  Note that in this case all audio frames are encoded
 using the same frame type, i.e., there is only one ToC entry.

Sjoberg, et al. Standards Track [Page 19] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

  0                   1
  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |0| Frame Type1 |  #frames = 3  |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 The next example depicts a ToC of three entries in basic mode.  Note
 that in this case the payload also carries three frames, but three
 ToC entries are needed because the frames of the payload are encoded
 using different frame types.
  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| Frame Type1 |  #frames = 1  |1| Frame Type2 |  #frames = 1  |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |0| Frame Type3 |  #frames = 1  |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 The following example illustrates a ToC with two entries in
 interleaved mode using four-bit displacement fields.  The payload
 includes two groups of frames, the first one including a single
 frame, and the other one consisting of two frames.
  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| Frame Type1 |  #frames = 1  |  DIS1 |  padd |0| Frame Type2 |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |  #frames = 2  |  DIS1 |  DIS2 |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

4.3.3. Audio Data

 Audio data of a payload consists of zero or more audio frames, as
 described in the ToC of the payload.
 ToC entries with FT=14 or 15 represent frame types with a length of
 0.  Hence, no data SHALL be placed in the audio data section to
 represent frames of this type.
 As already discussed, each audio frame of an extension frame type
 represents an AMR-WB+ transport frame corresponding to the encoding
 of 512 samples of audio, sampled with the internal sampling frequency
 specified by the ISF indicator.  As an exception, frame types with
 index 10-13 are only capable of using a single internal sampling
 frequency (25600 Hz).  The encoding rates (combination of core bit-
 rate and stereo bit-rate) are indicated in the frame type field of

Sjoberg, et al. Standards Track [Page 20] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

 the corresponding ToC entry.  The octet length of the audio frame is
 implicitly defined by the frame type field and is given in Tables 21
 and 25 of [1].  The order and numbering notation of the bits are as
 specified in [1].  For the AMR-WB+ extension frame types and comfort
 noise frames, the bits are in the order produced by the encoder.  The
 last octet of each audio frame MUST be padded with zeroes at the end
 if not all bits in the octet are used.  In other words, each audio
 frame MUST be octet-aligned.

4.3.4. Methods for Forming the Payload

 The payload begins with the payload header, followed by the table of
 contents, which consists of a list of ToC entries.
 The audio data follows the table of contents.  All the octets
 comprising an audio frame SHALL be appended to the payload as a unit.
 The audio frames are packetized in timestamp order within each group
 of frames (per ToC entry).  The groups of frames are packetized in
 the same order as their corresponding ToC entries.  Note that there
 are no data octets in a group having a ToC entry with FT=14 or FT=15.

4.3.5. Payload Examples

4.3.5.1. Example 1: Basic Mode Payload Carrying Multiple Frames Encoded

        Using the Same Frame Type
 Figure 4 depicts a payload that carries three AMR-WB+ frames encoded
 using 14 kbit/s frame type (FT=26) with a frame length of 280 bits
 (35 bytes).  The internal sampling frequency in this example is 25.6
 kHz (ISF = 8).  The TFI for the first frame is 2, indicating that the
 first transport frame in this payload is the third in a super-frame.
 Since this payload is in the basic mode, the subsequent frames of the
 payload are consecutive frames in decoding order, i.e., the fourth
 transport frame of the current super-frame and the first transport
 frame of the next super-frame.  Note that because the frames are all
 encoded using the same frame type, only one ToC entry is required.

Sjoberg, et al. Standards Track [Page 21] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

  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
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | ISF = 8 | 2 |0|0|  FT = 26    |  #frames = 3  |   f1(0...7)   |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 : ...                                                           :
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | ...           | f1(272...279) |   f2(0...7)   |               |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 : ...                                                           :
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | f2(272...279) |   f3(0...7)   | ...                           |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 : ...                                                           :
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | ...                                           | f3(272...279) |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 Figure 4: An example of a basic mode payload carrying three frames
           of the same frame type

4.3.5.2. Example 2: Basic Mode Payload Carrying Multiple Frames Encoded

        Using Different Frame Types
 Figure 5 depicts a payload that carries three AMR-WB+ frames; the
 first frame is encoded using 18.4 kbit/s frame type (FT=33) with a
 frame length of 368 bits (46 bytes), and the two subsequent frames
 are encoded using 20 kbit/s frame type (FT=35) having frame length of
 400 bits (50 bytes).  The internal sampling frequency in this example
 is 32 kHz (ISF = 10), implying the overall bit-rates of 23 kbit/s for
 the first frame of the payload, and 25 kbit/s for the subsequent
 frames.  The TFI for the first frame is 3, indicating that the first
 transport frame in this payload is the fourth in a super-frame.
 Since this is a payload in the basic mode, the subsequent frames of
 the payload are consecutive frames in decoding order, i.e., the first
 and second transport frames of the current super-frame.  Note that
 since the payload carries two different frame types, there are two
 ToC entries.

Sjoberg, et al. Standards Track [Page 22] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

  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
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |  ISF=10 | 3 |0|1|  FT = 33    |  #frames = 1  |0|  FT = 35    |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |  #frames = 2  |   f1(0...7)   | ...                           |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 : ...                                                           :
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | ...                           | f1(360...367) |   f2(0...7)   |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 : ...                                                           :
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | f2(392...399) |   f3(0...7)   | ...                           |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 : ...                                                           :
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | ...                           | f3(392...399) |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 Figure 5: An example of a basic mode payload carrying three frames
           employing two different frame types

4.3.5.3. Example 3: Payload in Interleaved Mode

 The example in Figure 6 depicts a payload in interleaved mode,
 carrying four frames encoded using 32 kbit/s frame type (FT=47) with
 frame length of 640 bits (80 bytes).  The internal sampling frequency
 is 38.4 kHz (ISF = 13), implying a bit-rate of 48 kbit/s for all
 frames in the payload.  The TFI for the first frame is 0; hence, it
 is the first transport frame of a super-frame.  The displacement
 fields for the subsequent frames are DIS2=18, DIS3=15, and DIS4=10,
 which indicates that the subsequent frames have the TFIs of 3, 3, and
 2, respectively.  The long displacement field flag L in the payload
 header is set to 1, which results in the use of eight bits for the
 displacement fields in the ToC entry.  Note that since all frames of
 this payload are encoded using the same frame type, there is need
 only for a single ToC entry.  Furthermore, the displacement field for
 the first frame (corresponding to the first ToC entry with DIS1=0)
 must be ignored, since its timestamp and TFI are defined by the RTP
 timestamp and the TFI found in the payload header.
 The RTP timestamp values of the frames in this example are:
 Frame1: TS1 = RTP Timestamp
 Frame2: TS2 = TS1 + 19 * 960
 Frame3: TS3 = TS2 + 16 * 960
 Frame4: TS4 = TS3 + 11 * 960

Sjoberg, et al. Standards Track [Page 23] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

  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
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |  ISF=13 | 0 |1|0|  FT = 47    |  #frames = 4  |   DIS1 = 0    |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |   DIS2 = 18   |   DIS3 = 15   |   DIS4 = 10   |   f1(0...7)   |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 : ...                                                           :
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | ...                           | f1(632...639) |   f2(0...7)   |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 : ...                                                           :
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | ...                           | f2(632...639) |   f3(0...7)   |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 : ...                                                           :
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | ...                           | f3(632...639) |   f4(0...7)   |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 : ...                                                           :
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | ...                           | f4(632...639) |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 Figure 6: An example of an interleaved mode payload carrying four
           frames at the same frame type

4.4. Interleaving Considerations

 The use of interleaving requires further considerations.  As
 presented in the example in Section 3.6.2, a given interleaving
 pattern requires a certain amount of the deinterleaving buffer.  This
 buffer space, expressed in a number of transport frame slots, is
 indicated by the "interleaving" media type parameter.  The number of
 frame slots needed can be converted into actual memory requirements
 by considering the 80 bytes per frame used by the largest combination
 of AMR-WB+'s core and stereo rates.
 The information about the frame buffer size is not always sufficient
 to determine when it is appropriate to start consuming frames from
 the interleaving buffer.  There are two cases in which additional
 information is needed: first, when switching of the ISF occurs, and
 second, when the interleaving pattern changes.  The "int-delay" media
 type parameter is defined to convey this information.  It allows a
 sender to indicate the minimal media time that needs to be present in
 the buffer before the decoder can start consuming frames from the
 buffer.  Because the sender has full control over ISF changes and the
 interleaving pattern, it can calculate this value.

Sjoberg, et al. Standards Track [Page 24] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

 In certain cases (for example, if joining a multicast session with
 interleaving mid-session), a receiver may initially receive only part
 of the packets in the interleaving pattern.  This initial partial
 reception (in frame sequence order) of frames can yield too few
 frames for acceptable quality from the audio decoding.  This problem
 also arises when using encryption for access control, and the
 receiver does not have the previous key.
 Although the AMR-WB+ is robust and thus tolerant to a high random
 frame erasure rate, it would have difficulties handling consecutive
 frame losses at startup.  Thus, some special implementation
 considerations are described.  In order to handle this type of
 startup efficiently, it must be noted that decoding is only possible
 to start at the beginning of a super-frame, and that holds true even
 if the first transport frame is indicated as lost.  Secondly,
 decoding is only RECOMMENDED to start if at least 2 transport frames
 are available out of the 4 belonging to that super-frame.
 After receiving a number of packets, in the worst case as many
 packets as the interleaving pattern covers, the previously described
 effects disappear and normal decoding is resumed.
 Similar issues arise when a receiver leaves a session or has lost
 access to the stream.  If the receiver leaves the session, this would
 be a minor issue since playout is normally stopped.  It is also a
 minor issue for the case of lost access, since the AMR-WB+ error
 concealment will fade out the audio if massive consecutive losses are
 encountered.
 The sender can avoid this type of problem in many sessions by
 starting and ending interleaving patterns correctly when risks of
 losses occur.  One such example is a key-change done for access
 control to encrypted streams.  If only some keys are provided to
 clients and there is a risk of their receiving content for which they
 do not have the key, it is recommended that interleaving patterns not
 overlap key changes.

4.5. Implementation Considerations

 An application implementing this payload format MUST understand all
 the payload parameters.  Any mapping of the parameters to a signaling
 protocol MUST support all parameters.  So an implementation of this
 payload format in an application using SDP is required to understand
 all the payload parameters in their SDP-mapped form.  This
 requirement ensures that an implementation always can decide whether
 it is capable of communicating.

Sjoberg, et al. Standards Track [Page 25] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

 Both basic and interleaved mode SHALL be implemented.  The
 implementation burden of both is rather small, and requiring both
 ensures interoperability.  As the AMR-WB+ codec contains the full
 functionality of the AMR-WB codec, it is RECOMMENDED to also
 implement the payload format in RFC 3267 [7] for the AMR-WB frame
 types when implementing this specification.  Doing so makes
 interoperability with devices that only support AMR-WB more likely.
 The switching of ISF, when combined with packet loss, could result in
 concealment using the wrong audio frame length.  This can occur if
 packet losses result in lost frames directly after the point of ISF
 change.  The packet loss would prevent the receiver from noticing the
 changed ISF and thereby conceal the lost transport frame with the
 previous ISF, instead of the new one.  Although always later
 detectable, such an error results in frame boundary misalignment,
 which can cause audio distortions and problems with synchronization,
 as too many or too few audio samples were created.  This problem can
 be mitigated in most cases by performing ISF recovery prior to
 concealment as outlined in Section 4.5.1.

4.5.1. ISF Recovery in Case of Packet Loss

 In case of packet loss, it is important that the AMR-WB+ decoder
 initiates a proper error concealment to replace the frames carried in
 the lost packet.  A loss concealment algorithm requires a codec
 framing that matches the timestamps of the correctly received frames.
 Hence, it is necessary to recover the timestamps of the lost frames.
 Doing so is non-trivial because the codec frame length that is
 associated with the ISF may have changed during the frame loss.
 In the following, the recovery of the timestamp information of lost
 frames is illustrated by the means of an example.  Two frames with
 timestamps t0 and t1 have been received properly, the first one being
 the last packet before the loss, and the latter one being the first
 packet after the loss period.  The ISF values for these packets are
 isf0 and isf1, respectively.  The TFIs of these frames are tfi0 and
 tfi1, respectively.  The associated frame lengths (in timestamp
 ticks) are given as L0 and L1, respectively.  In this example three
 frames with timestamps x1 - x3 have been lost.  The example further
 assumes that ISF changes once from isf0 to isf1 during the frame loss
 period, as shown in the figure below.
 Since not all information required for the full recovery of the
 timestamps is generally known in the receiver, an algorithm is needed
 to estimate the ISF associated with the lost frames.  Also, the
 number of lost frames needs to be recovered.

Sjoberg, et al. Standards Track [Page 26] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

   |<---L0--->|<---L0--->|<-L1->|<-L1->|<-L1->|
   |   Rxd    |   lost   | lost | lost |  Rxd |
 --+----------+----------+------+------+------+--
   t0         x1         x2     x3     t1
 Example Algorithm:
 Start:                              # check for frame loss
 If (t0 + L0) == t1 Then goto End    # no frame loss
 Step 1:                             # check case with no ISF change
 If (isf0 != isf1) Then goto Step 2  # At least one ISF change
 If (isFractional(t1 - t0)/L0) Then goto Step 3
                                     # More than 1 ISF change
 Return recovered timestamps as
 x(n) = t0 + n*L1 and associated ISF equal to isf0,
 for 0 < n < (t1 - t0)/L0
 goto End
 Step 2:
 Loop initialization: n := 4 - tfi0 mod 4
 While n <= (t1-t0)/L0
   Evaluate m := (t1 - t0 - n*L0)/L1
   If (isInteger(m) AND ((tfi0+n+m) mod 4 == tfi1)) Then goto found;
   n := n+4
   endloop
 goto step 3                         # More than 1 ISF change
 found:
 Return recovered timestamps and ISFs as
 x(i) = t0 + i*L0 and associated ISF equal to isf0, for 0 < i <= n
 x(i) = t0 + n*L0 + (i-n)*L1 and associated ISF equal to isf1,
 for n < i <= n+m
 goto End
 Step 3:
 More than 1 ISF change has occurred.  Since ISF changes can be
 assumed to be infrequent, such a situation occurs only if long
 sequences of frames are lost.  In that case it is probably not useful
 to try to recover the timestamps of the lost frames.  Rather, the
 AMR-WB+ decoder should be reset, and decoding should be resumed
 starting with the frame with timestamp t1.
 End:

Sjoberg, et al. Standards Track [Page 27] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

 The above algorithm still does not solve the issue when the receiver
 buffer depth is shallower than the loss burst.  In this kind of case,
 where the concealment must be done without any knowledge about future
 frames, the concealment may result in loss of frame boundary
 alignment.  If that occurs, it may be necessary to reset and restart
 the codec to perform resynchronization.

4.5.2. Decoding Validation

 If the receiver finds a mismatch between the size of a received
 payload and the size indicated by the ToC of the payload, the
 receiver SHOULD discard the packet.  This is recommended because
 decoding a frame parsed from a payload based on erroneous ToC data
 could severely degrade the audio quality.

5. Congestion Control

 The general congestion control considerations for transporting RTP
 data apply; see RTP [3] and any applicable RTP profile like AVP [9].
 However, the multi-rate capability of AMR-WB+ audio coding provides a
 mechanism that may help to control congestion, since the bandwidth
 demand can be adjusted (within the limits of the codec) by selecting
 a different coding frame type or lower internal sampling rate.
 The number of frames encapsulated in each RTP payload highly
 influences the overall bandwidth of the RTP stream due to header
 overhead constraints.  Packetizing more frames in each RTP payload
 can reduce the number of packets sent and hence the header overhead,
 at the expense of increased delay and reduced error robustness.
 If forward error correction (FEC) is used, the amount of FEC-induced
 redundancy needs to be regulated such that the use of FEC itself does
 not cause a congestion problem.

6. Security Considerations

 RTP packets using the payload format defined in this specification
 are subject to the general security considerations discussed in RTP
 [3] and any applicable profile such as AVP [9] or SAVP [10].  As this
 format transports encoded audio, the main security issues include
 confidentiality, integrity protection, and data origin authentication
 of the audio itself.  The payload format itself does not have any
 built-in security mechanisms.  Any suitable external mechanisms, such
 as SRTP [10], MAY be used.

Sjoberg, et al. Standards Track [Page 28] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

 This payload format and the AMR-WB+ decoder do not exhibit any
 significant non-uniformity in the receiver-side computational
 complexity for packet processing, and thus are unlikely to pose a
 denial-of-service threat due to the receipt of pathological data.

6.1. Confidentiality

 In order to ensure confidentiality of the encoded audio, all audio
 data bits MUST be encrypted.  There is less need to encrypt the
 payload header or the table of contents since they only carry
 information about the frame type.  This information could also be
 useful to a third party, for example, for quality monitoring.
 The use of interleaving in conjunction with encryption can have a
 negative impact on confidentiality, for a short period of time.
 Consider the following packets (in brackets) containing frame numbers
 as indicated: {10, 14, 18}, {13, 17, 21}, {16, 20, 24} (a popular
 continuous diagonal interleaving pattern).  The originator wishes to
 deny some participants the ability to hear material starting at time
 16.  Simply changing the key on the packet with the timestamp at or
 after 16, and denying that new key to those participants, does not
 achieve this; frames 17, 18, and 21 have been supplied in prior
 packets under the prior key, and error concealment may make the audio
 intelligible at least as far as frame 18 or 19, and possibly further.

6.2. Authentication and Integrity

 To authenticate the sender of the speech, an external mechanism MUST
 be used.  It is RECOMMENDED that such a mechanism protects both the
 complete RTP header and the payload (speech and data bits).
 Data tampering by a man-in-the-middle attacker could replace audio
 content and also result in erroneous depacketization/decoding that
 could lower the audio quality.

7. Payload Format Parameters

 This section defines the parameters that may be used to select
 features of the AMR-WB+ payload format.  The parameters are defined
 as part of the media type registration for the AMR-WB+ audio codec.
 A mapping of the parameters into the Session Description Protocol
 (SDP) [6] is also provided for those applications that use SDP.
 Equivalent parameters could be defined elsewhere for use with control
 protocols that do not use MIME or SDP.
 The data format and parameters are only specified for real-time
 transport in RTP.

Sjoberg, et al. Standards Track [Page 29] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

7.1. Media Type Registration

 The media type for the Extended Adaptive Multi-Rate Wideband
 (AMR-WB+) codec is allocated from the IETF tree, since AMR-WB+ is
 expected to be a widely used audio codec in general streaming
 applications.
 Note: Parameters not listed below MUST be ignored by the receiver.
 Media Type name:     audio
 Media subtype name:  AMR-WB+
 Required parameters:
 None
 Optional parameters:
 channels:       The maximum number of audio channels used by the
                 audio frames.  Permissible values are 1 (mono) or 2
                 (stereo).  If no parameter is present, the maximum
                 number of channels is 2 (stereo).  Note: When set to
                 1, implicitly the stereo frame types cannot be used.
 interleaving:   Indicates that interleaved mode SHALL
                 be used for the payload.  The parameter specifies
                 the number of transport frame slots required in a
                 deinterleaving buffer (including the frame that is
                 ready to be consumed).  Its value is equal to one
                 plus the maximum number of frames that precede any
                 frame in transmission order and follow the frame in
                 RTP timestamp order.  The value MUST be greater than
                 zero.  If this parameter is not present,
                 interleaved mode SHALL NOT be used.
 int-delay:      The minimal media time delay in RTP timestamp ticks
                 that is needed in the deinterleaving buffer, i.e.,
                 the difference in RTP timestamp ticks between the
                 earliest and latest audio frame present in the
                 deinterleaving buffer.
 ptime:          See Section 6 in RFC 2327 [6].
 maxptime:       See Section 8 in RFC 3267 [7].
 Restriction on Usage:
              This type is only defined for transfer via RTP (STD 64).

Sjoberg, et al. Standards Track [Page 30] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

 Encoding considerations:
              An RTP payload according to this format is binary data
              and thus may need to be appropriately encoded in non-
              binary environments.  However, as long as used within
              RTP, no encoding is necessary.
 Security considerations:
              See Section 6 of RFC 4352.
 Interoperability considerations:
              To maintain interoperability with AMR-WB-capable end-
              points, in cases where negotiation is possible and the
              AMR-WB+ end-point supporting this format also supports
              RFC 3267 for AMR-WB transport, an AMR-WB+ end-point
              SHOULD declare itself also as AMR-WB capable (i.e.,
              supporting also "audio/AMR-WB" as specified in RFC
              3267).
              As the AMR-WB+ decoder is capable of performing stereo
              to mono conversions, all receivers of AMR-WB+ should be
              able to receive both stereo and mono, although the
              receiver is only capable of playout of mono signals.
 Public specification:
              RFC 4352
              3GPP TS 26.290, see reference [1] of RFC 4352
 Additional information:
              This MIME type is not applicable for file storage.
              Instead, file storage of AMR-WB+ encoded audio is
              specified within the 3GPP-defined ISO-based multimedia
              file format defined in 3GPP TS 26.244; see reference
              [14] of RFC 4352.  This file format has the MIME types
              "audio/3GPP" or "video/3GPP" as defined by RFC 3839
              [15].
 Person & email address to contact for further information:
              magnus.westerlund@ericsson.com
              ari.lakaniemi@nokia.com
 Intended usage: COMMON.
              It is expected that many IP-based streaming
              applications will use this type.
 Change controller:
              IETF Audio/Video Transport working group delegated from
              the IESG.

Sjoberg, et al. Standards Track [Page 31] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

7.2. 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)
 [6], which is commonly used to describe RTP sessions.  When SDP is
 used to specify an RTP session using this RTP payload format, the
 mapping is as follows:
  1. The media type ("audio") is used in SDP "m=" as the media name.
  1. The media type (payload format name) is used in SDP "a=rtpmap" as

the encoding name. The RTP clock rate in "a=rtpmap" SHALL be

    72000 for AMR-WB+, and the encoding parameter number of channels
    MUST either be explicitly set to 1 or 2, or be omitted, implying
    the default value of 2.
  1. The parameters "ptime" and "maxptime" are placed in the SDP

attributes "a=ptime" and "a=maxptime", respectively.

  1. Any remaining parameters are placed in the SDP "a=fmtp" attribute

by copying them directly from the MIME media type string as a

    semicolon-separated list of parameter=value pairs.

7.2.1. Offer-Answer Model Considerations

 To achieve good interoperability in an Offer-Answer [8] negotiation
 usage, the following considerations should be taken into account:
 For negotiable offer/answer usage the following interpretation rules
 SHALL be applied:
  1. The "interleaving" parameter is symmetric, thus requiring that the

answerer must also include it for the answer to an offered payload

    type that contains the parameter.  However, the buffer space value
    is declarative in usage in unicast.  For multicast usage, the same
    value in the response is required in order to accept the payload
    type.  For streams declared as sendrecv or recvonly: The receiver
    will accept reception of streams using the interleaved mode of the
    payload format.  The value declares the amount of buffer space the
    receiver has available for the sender to utilize.  For sendonly
    streams, the parameter indicates the desired configuration and
    amount of buffer space.  An answerer is RECOMMENDED to respond
    using the offered value, if capable of using it.

Sjoberg, et al. Standards Track [Page 32] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

  1. The "int-delay" parameter is declarative. For streams declared as

sendrecv or recvonly, the value indicates the maximum initial

    delay the receiver will accept in the deinterleaving buffer.  For
    sendonly streams, the value is the amount of media time the sender
    desires to use.  The value SHOULD be copied into any response.
  1. The "channels" parameter is declarative. For "sendonly" streams,

it indicates the desired channel usage, stereo and mono, or mono

    only.  For "recvonly" and "sendrecv" streams, the parameter
    indicates what the receiver accepts to use.  As any receiver will
    be capable of receiving stereo frame type and perform local mixing
    within the AMR-WB+ decoder, there is normally only one reason to
    restrict to mono only: to avoid spending bit-rate on data that are
    not utilized if the front-end is only capable of mono.
  1. The "ptime" parameter works as indicated by the offer/answer model

[8]; "maxptime" SHALL be used in the same way.

  1. To maintain interoperability with AMR-WB in cases where

negotiation is possible, an AMR-WB+ capable end-point that also

    implements the AMR-WB payload format [7] is RECOMMENDED to declare
    itself capable of AMR-WB as it is a subset of the AMR-WB+ codec.
 In declarative usage, like SDP in RTSP [16] or SAP [17], the
 following interpretation of the parameters SHALL be done:
  1. The "interleaving" parameter, if present, configures the payload

format in that mode, and the value indicates the number of frames

    that the deinterleaving buffer is required to support to be able
    to handle this session correctly.
  1. The "int-delay" parameter indicates the initial buffering delay

required to receive this stream correctly.

  1. The "channels" parameter indicates if the content being

transmitted can contain either both stereo and mono rates, or only

    mono.
  1. All other parameters indicate values that are being used by the

sending entity.

Sjoberg, et al. Standards Track [Page 33] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

7.2.2. Examples

 One example of an SDP session description utilizing AMR-WB+ mono and
 stereo encoding follows.
  m=audio 49120 RTP/AVP 99
  a=rtpmap:99 AMR-WB+/72000/2
  a=fmtp:99 interleaving=30; int-delay=86400
  a=maxptime:100
 Note that the payload format (encoding) names are commonly shown in
 uppercase.  Media subtypes are commonly shown in lowercase.  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.

8. IANA Considerations

 The IANA has registered one new MIME subtype (audio/amr-wb+); see
 Section 7.

9. Contributors

 Daniel Enstrom has contributed in writing the codec introduction
 section.  Stefan Bruhn has contributed by writing the ISF recovery
 algorithm.

10. Acknowledgements

 The authors would like to thank Redwan Salami and Stefan Bruhn for
 their significant contributions made throughout the writing and
 reviewing of this document.  Dave Singer contributed by reviewing and
 suggesting improved language.  Anisse Taleb and Ingemar Johansson
 contributed by implementing the payload format and thus helped locate
 some flaws.  We would also like to acknowledge Qiaobing Xie, coauthor
 of RFC 3267, on which this document is based.

Sjoberg, et al. Standards Track [Page 34] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

11. References

11.1. Normative References

 [1]  3GPP TS 26.290 "Audio codec processing functions; Extended
      Adaptive Multi-Rate Wideband (AMR-WB+) codec; Transcoding
      functions", version 6.3.0 (2005-06), 3rd Generation Partnership
      Project (3GPP).
 [2]  Bradner, S., "Key words for use in RFCs to Indicate Requirement
      Levels", BCP 14, RFC 2119, March 1997.
 [3]  Schulzrinne, H.,  Casner, S., Frederick, R., and V. Jacobson,
      "RTP: A Transport Protocol for Real-Time Applications", STD 64,
      RFC 3550, July 2003.
 [4]  3GPP TS 26.192 "AMR Wideband speech codec; Comfort Noise
      aspects", version 6.0.0 (2004-12), 3rd Generation Partnership
      Project (3GPP).
 [5]  3GPP TS 26.193 "AMR Wideband speech codec; Source Controlled
      Rate operation", version 6.0.0 (2004-12), 3rd Generation
      Partnership Project (3GPP).
 [6]  Handley, M. and V. Jacobson, "SDP: Session Description
      Protocol", RFC 2327, April 1998.
 [7]  Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie, "Real-
      Time Transport Protocol (RTP) Payload Format and File Storage
      Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate
      Wideband (AMR-WB) Audio Codecs", RFC 3267, June 2002.
 [8]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
      Session Description Protocol (SDP)", RFC 3264, June 2002.
 [9]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video
      Conferences with Minimal Control", STD 65, RFC 3551, July 2003.

11.2. Informative References

 [10] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
      Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC
      3711, March 2004.
 [11] Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format for
      Generic Forward Error Correction", RFC 2733, December 1999.

Sjoberg, et al. Standards Track [Page 35] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

 [12] 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.
 [13] 3GPP TS 26.233 "Packet Switched Streaming service", version
      5.7.0 (2005-03), 3rd Generation Partnership Project (3GPP).
 [14] 3GPP TS 26.244 "Transparent end-to-end packet switched streaming
      service (PSS); 3GPP file format (3GP)", version 6.4.0 (2005-09),
      3rd Generation Partnership Project (3GPP).
 [15] Castagno, R. and D. Singer, "MIME Type Registrations for 3rd
      Generation Partnership Project (3GPP) Multimedia files", RFC
      3839, July 2004.
 [16] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time Streaming
      Protocol (RTSP)", RFC 2326, April 1998.
 [17] Handley, M., Perkins, C., and E. Whelan, "Session Announcement
      Protocol", RFC 2974, October 2000.
 [18] 3GPP TS 26.140 "Multimedia Messaging Service (MMS); Media
      formats and codes", version 6.2.0 (2005-03), 3rd Generation
      Partnership Project (3GPP).
 [19] 3GPP TS 26.140 "Multimedia Broadcast/Multicast Service (MBMS);
      Protocols and codecs", version 6.3.0 (2005-12), 3rd Generation
      Partnership Project (3GPP).
 Any 3GPP document can be downloaded from the 3GPP webserver,
 "http://www.3gpp.org/", see specifications.

Sjoberg, et al. Standards Track [Page 36] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

Authors' Addresses

 Johan Sjoberg
 Ericsson Research
 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
 Stephan Wenger
 Nokia Corporation
 P.O. Box 100
 FIN-33721 Tampere
 FINLAND
 Phone: +358-50-486-0637
 EMail: Stephan.Wenger@nokia.com

Sjoberg, et al. Standards Track [Page 37] RFC 4352 RTP Payload Format for AMR-WB+ January 2006

Full Copyright Statement

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 This document is subject to the rights, licenses and restrictions
 contained in BCP 78, and except as set forth therein, the authors
 retain all their rights.
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Sjoberg, et al. Standards Track [Page 38]

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