Working Group AVT                                              C. Hoene
Internet Draft                                  University of Tuebingen
Intended status: Standards Track                             F. de Bont
Expires: December 2010 June 2011                                  Philips Electronics
                                                           June 9,
                                                      December 11, 2010

            RTP Payload Format for Bluetooth's SBC audio codec

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   This document specifies a Real-time Transport Protocol (RTP) payload
   format to be used for the low complexity subband codec (SBC), which
   is the mandatory audio codec of the Advanced Audio Distribution
   Profile (A2DP) Specification written by the Bluetooth(r) Special
   Interest Group (SIG). The payload format is designed to be able to
   interoperate with existing Bluetooth A2DP devices, to provide high
   streaming audio quality, interactive audio transmission over the
   internet, and ultra-low delay coding for jam sessions on the
   internet. This document contains also a media type registration which
   specifies the use of the RTP payload format.

Table of Contents

   1. Introduction...................................................3
   2. Conventions used in this document..............................3
   3. Background.....................................................3
   4. Usage Scenarios................................................5
      4.1. Scenario 1: Interconnection of A2DP devices...............5
      4.2. Scenario 2: High quality interactive audio transmissions..6
      4.3. Scenario 3: Ensembles performing over a network...........6
   5. Header Usage...................................................7
   6. Payload Format.................................................8
      6.1. Media payload format header...............................8
      6.2. SBC Frame Structure.......................................9
      6.3. Frame header..............................................9
      6.4. Remaining frame..........................................12
   7. Payload Format Parameters.....................................12
      7.1. SBC Media Type Registration..............................12
         7.1.1. Capabilities: A2DP modes............................13
         7.1.2. Capabilities: other modes...........................15
      7.2. Mapping to SDP Parameters................................15
         7.2.1. Offer-Answer Model Considerations...................15
         7.2.2. Declarative SDP Considerations......................17
   8. Congestion Control............................................17
   9. Packet loss concealment.......................................18
   10. Security Considerations......................................19
   11. IANA Considerations..........................................19
   12. References...................................................20
      12.1. Normative References....................................20
      12.2. Informative References..................................20
   13. Acknowledgments..............................................22

1. Introduction

   The Bluetooth(r) Special Interest Group (SIG) specifies in the
   Advanced Audio Distribution Profile (A2DP) [A2DPV10] a mono and
   stereo high quality audio subband codec (SBC). This document
   specifies the payload format for the encapsulation of SBC encoded
   audio frames into the Real-time Transport Protocol (RTP).

   SBC has a low computational complexity at modest compression rates.
   Its bit rate can be controlled widely. Recommended operational modes
   range from 127 to 345 kb/s, for mono and stereo audio signals. SBC's
   algorithmic delay can be as low as 16 samples making it ideal for
   ensembles playing music over the network requiring ultra low acoustic

2. Conventions used in this document

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in RFC-2119 [RFC2119].

   The following acronyms are used in this document:

     A2DP   - Audio Distribution Profile
     AAC    - Advanced Audio Coding
     ATRAC  - Adaptive Transform Acoustic Coding
     DCCP   - Datagram Congestion Control Protocol
     MP3    - MPEG-1 Audio Layer 3
     SBC    - SubBand Codec
     SIG    - Special Interest Group

3. Background

   The A2DP specification is intended for streaming of music content to
   headphones, headsets, or speakers over Bluetooth wireless channels.
   A2DP supports multiple audio coding including MP3, AAC, ATRAC, which
   are all non-mandatory. To ensure interoperability, the SBC codec has
   been specified, which shall be included into all A2DP Bluetooth

   The SBC is a low complexity subband codec based on earlier work
   presented in [Bon1995] and [Rault1989]. It has a moderate compression
   ratio. The SBC encoder has filter banks splitting the audio signal
   into 4 or 8 subbands. Then the codec decides with how many bits each
   subband is encoded and finally quantizes the subband signals
   blockwise. An SBC frame can have different block sizes. The size of a
   block can be 4, 8, 12 or 16. Both decoder and encoder shall support
   all four block sizes.

   SBC can operate at four different sampling frequencies. The sampling
   frequency can be selected from a set of 16, 32, 44.1, and 48 kHz. It
   is mandatory that each SBC decoder can operate at the frequencies
   44.1 and 48 kHz. Each SBC encoder shall work at least at a sampling
   rate of 44.1 or 48 kHz.

   Four channel modes are supported, which are mono, dual channel,
   stereo, and joint-stereo. The decoder shall support all four of them;
   the encoder shall support mono and at least one additional mode.

   SBC can use four or eight subbands. The decoder shall support both;
   the encoder shall support at least 8 subbands.

   The bit allocation modes of SBC can be either based on signal to
   noise ratio or on loudness. The decoder shall support both modes; the
   encoder shall support at least the loudness mode.

   The SBC encoder reduces one block to a given number of bits. The bit-
   pool variable defines how many bits are used per block. A2DP devices
   define the range of valid bit-pool values by providing minimum and
   maximum bit-pool values. The bit-pool values shall range from 2 to
   250 but shall not be larger than number of subbands times 16 for the
   mono and dual and times 32 for the stereo and joint-stereo channel

   SBC encoders inside A2DP devices may be capable of changing the bit-
   pool parameter dynamically during the encoding process. For example,
   algorithms were invented that change the number of bits depending on
   the current acoustic content [Pilati2008].

   The decoder shall support all possible bit-pool values that do not
   result in excess of maximum bit rate, which is 320kb/s for mono and
   512kb/s for two-channel modes. The encoder is required to support at
   least one possible bit-pool value. The A2DP specification recommends
   the encoding parameters given in Table 1.

   | SBC encoder settings at Medium Quality                     |
   |                                |    Mono     | Joint Stereo|
   | Sampling frequency (kHz)       | 44.1 |  48  | 44.1 |  48  |
   | Bitpool value                  |  19  |  18  |  35  |  33  |
   | Resulting frame length (bytes) |  46  |  44  |  83  |  79  |
   | Resulting bit rate (kb/s)      | 127  | 132  | 229  | 237  |
   | SBC encoder settings at High Quality                       |
   |                                |    Mono     | Joint Stereo|
   | Sampling frequency (kHz)       | 44.1 |  48  | 44.1 |  48  |
   | Bitpool value                  |  31  |  29  |  53  |  51  |
   | Resulting frame length (bytes) |  70  |  66  | 119  | 115  |
   | Resulting bit rate (kb/s)      | 193  | 198  | 328  | 345  |
   + Other settings: Block length = 16, loudness, subbands = 8  |

   Table 1: Recommended sets of SBC parameters in the SRC device as
   given in [A2DPV10]

   The A2DP V1.0 specification describes a media payload format, which
   we adopt in this document one-to-one without any change.

4. Usage Scenarios

   As compared to many other encoding schemes, the SBC is general enough
   to support multiple, quite diverse usage scenarios. Thus, it might be
   required to change the behavior of the encoding and transmission to
   achieve a good performance for a given usage scenario. Thus, we
   enlist three main scenarios and describe their quality requirements
   and their impact on the encoding and transmission.

4.1. Scenario 1: Interconnection of A2DP devices

   In this scenario it is intended to interconnect Bluetooth A2DP
   devices. RTP frames generated by an A2DP device can be transmitted
   directly via this RTP profile. Vice versa, an A2DP device should be
   able to receive the RTP profile by default. Thus, the payload format
   describe in this RFC MUST be fully interoperable with any A2DP

   The transmission between two A2DP devices has a constant frame rate
   with a sender-controlled bit rate. It is not anticipated that the
   transmission is adapted to congestion and bandwidth variation.

4.2. Scenario 2: High quality interactive audio transmissions

   In the second scenario we consider a telephone call having a very
   good audio quality at modest acoustic one-way latencies ranging from
   50 and 150 ms [ITUG107], so that music can be listened over the
   telephone while two persons talk together interactively.

   In addition, the reliability of the audio transmission should be
   high, even in cases of low and varying bandwidth.

   This second scenario assumes that the SBC transmission is used on top
   of a transport protocol that implements a congestion control
   algorithm. Using the SBC encoding, the sampling, bit, and frame rates
   should be controlled to cope with congestion. For example, if the
   available transmission bandwidth is too low to allow SBC to transmit
   audio at a high quality, the application can lower the sampling, bit,
   or frame rate of the stream at the cost of higher algorithmic delay
   or a degraded audio quality. In this case, changing the sampling or
   frame rate may cause a short acoustic artifact because SBC's internal
   filters must be reset.

   The A2DP media format does not allow a dynamic change of the encoding
   parameters beside the bit-pool value. The encoding parameters can
   only be altered with the "Change Parameters" procedure, which is
   defined in [GAVDPV12].  Such a change will cause a hearable
   interruption and thus shall be avoided.

   If an application using RTP wants to switch between different sets of
   encoding parameters, then these set of parameter CAN be either
   negotiate beforehand (as described in Section 7.2.) or an
   renegotiation similar to the "Change Parameters" procedure CAN take
   place. An application MUST NOT change the sampling frequency, block
   length, encoding mode or the number of subbands within one RTP
   session having the same RTP payload identifier.

4.3. Scenario 3: Ensembles performing over a network

   In some usage scenarios, users want to act simultaneously and not
   just interactively. For example, if persons sing in a chorus, if
   musicians jam, or if e-sportsmen play computer games in a team
   together, they need to acoustically communicate.

   In these scenarios, the latency requirements are much harder than for
   interactive usages. For example, if two musicians are placed more
   than 10 meters apart, they can hardly keep synchronized. Empirical
   studies [Gurevich2004] have shown that if ensembles playing over
   networks, the optimal acoustic latency is around 11.5 ms with
   targeted range from 10 to 25 ms.

   To fulfill such requirements, it might be necessary to further reduce
   the algorithmic coding delay by varying the block length parameter.
   The default value of the block length parameter is chosen such that
   the coding efficiency is maximized. For example, at 44.1 kHz and
   using 8 subbands and a block length of 16, the algorithmic delay is
   4.72 ms (208 samples). The value of the block length parameter can be
   decreased, at the expense of a higher bit rate or lower quality, to
   lower the latency to fulfill the very stringent latency requirements
   of this scenario.

   Still, given the speed of light as the fundamental limit of speed of
   information exchange, distributed ensembles can perform only
   regionally if latency budget of 25 ms must keep. Typically, an
   optical fiber has a refractive index of 1.46 and thus in an optical
   fiber bits travel about 5136 km one-way in 25 ms.

5. Header Usage

   The format of the RTP header is specified in [RFC3550]. The payload
   format defined in this document uses the fields of the header in a
   manner fully consistent with that specification.

   marker (M): In accordance with [A2DPV10] the marker bit MUST be set
             to zero.

   payload type (PT): The assignment of an RTP payload type for this
             packet format is outside the scope of the 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 codec or specify that the
             payload type is to be bound dynamically (see Section 6.2).

   timestamp (TS): The RTP timestamp clock frequency MUST be the same as
             the sampling frequency, which has been negotiated for the
             current RTP session (see Section 6.2). If a media payload
             consists of multiple SBC frames, the TS of the media packet
             header represents the TS of the first SBC frame. The TS of
             the following SBC frames MUST be calculated using the
             sampling rate and the number of samples per frame per
             channel. A change in sampling frequency MUST NOT occur
             within one media packet.
             A SBC frame may be fragmented into multiple media packets
             to reduce the packetisation delay. Then, all packets that
             make up a fragmented SBC frame MUST use the same TS.

6. Payload Format

   The format of the payload MUST follow exactly the description given
   in the appendix of [A2DPV10]. In the following, for the sake of
   clarity, we repeat the payload format definition.

   The payload MUST consist of one media payload format header described
   in Section 5.2 and SBC frames described in Section 5.3. Either an
   integral number of SBC frames or one fragment of an SBC frame can be

    (a) When the payload contains an integral number of SBC frames
   +--------+-----------+-----------   -+
   | Header | SBC frame | SBC frame ... |
   +--------+-----------+-----------   -+

   (b) When the SBC frame is fragmented
   | Header | First fragment of SBC frame           |

   | Header | Subsequent fragments of the SBC frame |

   A media payload always starts with an 8-bit header, which is placed
   before the SBC data.

   The SBC frame can be fragmented across several media payloads. All
   fragmented packets, except the last one, MUST have the same total
   data packet size.

   This payload fragmentation CAN be preferred against the fragmentation
   mechanisms of lower layers (e.g., IP) because the packetisation delay
   and thus the acoustic latency are reduced and the error robustness is
   increased because parts of the SBC frame can be considered for

6.1. Media payload format header

   The following figure shows the format of media payload header, which
   consists of one byte.

   0 1 2 3   4 5 6 7

   F bit - Set to 1 if the SBC frame is fragmented, otherwise set to 0.

   S bit - Set to 1 for the starting packet of a fragmented SBC frame,
             otherwise set to 0.

   L bit - Set to 1 for the last packet of a fragmented SBC frame,
             otherwise set to 0.

   RFA - SHOULD be zero, reserved for future addition.

   #frames (4 bits) - If the F bit is set to 0, this field indicates the
             number of frames contained in this packet. If the F bit is
             set to 1, this field indicates the number of remaining
             fragments, including the current fragment. Thus the last
             counter value MUST be one. For example, if there are three
             fragments then the counter has value 3, 2 and 1 for
             subsequent fragments.

6.2. SBC Frame Structure

   The complete SBC frame consists of a frame header, scale factors,
   audio samplings, and padding bits. The following diagram shows the
   general SBC frame format layout:

   | frame_header | scale_factors | audio_samples | padding |

   The following sections describe the audio format, which consists of
   bits stored in a bandwidth-efficient, compact mode.

6.3. Frame header

   The frame header consists of fields defined in [A2DPV10], which are
   fields and a RFA. The layout of the first four bytes of the frame
   header is given in the following table.

    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
   | SYNCWORD      |SF.|BL.|CM.|A|S|BITPOOL        |CRC_CHECK      |

   SYNCWORD (8 bits): The first field is the 8 bit synchronization word,
             which is always set to 156.

   SAMPLING_FREQUENCY (2 bits): The sampling frequency field indicates
             with which sampling frequency the SBC frame has been
             encoded. The table below specifies the corresponding
             sampling frequencies for the bit patterns. The sampling
             frequency MUST NOT be changed without changing the payload
             type, too.

   | SAMPLING_FREQUENCY | sampling       |
   |    bit 0 1         | frequency (Hz) |
   |        0 0         |      16000     |
   |        0 1         |      32000     |
   |        1 0         |      44100     |
   |        1 1         |      48000     |

   BLOCKS (2 bits): It indicates the block size with which the stream
             has been encoded. The block size is selected conforming to
             the table below. The block size MUST NOT be changed without
             changing the payload type, too.

   | BLOCKS  | Number of |
   | bit 0 1 | blocks    |
   |     0 0 |     4     |
   |     0 1 |     8     |
   |     1 0 |    12     |
   |     1 1 |    16     |

   CHANNEL_MODE (2 bits): These two bits indicate with which channel
             mode the frame has been encoded. The number of channels
             depends on this information. The channel mode MUST NOT be
             changed without changing the payload type, too.

   | CHANNEL_MODE | channel mode | number of |
   |    bit 0 1   |              | channels  |
   |        0 0   | MONO         |     1     |
   |        0 1   | DUAL_CHANNEL |     2     |
   |        1 0   | STEREO       |     2     |
   |        1 1   | JOINT_STEREO |     2     |

   ALLOCATION_METHOD (1 bit): This bit indicates how the bit pool is
             allocated to different subbands. Either it is based on the
             loudness of the sub band signal or on the signal to noise
             ratio. The allocation method MUST NOT be changed without
             changing the payload type, too.

   | ALLOCATION_METHOD | allocation |
   |       bit 0       | method     |
   |           0       |  LOUDNESS  |
   |           1       |     SNR    |

   SUBBANDS (1 bit): This bit indicates the number of subbands with
             which the frame has been encoded. The number of subband
             MUST NOT be changed without changing the payload type, too.

   | SUBBANDS | number of |
   |   bit 0  | subbands  |
   |       0  |      4    |
   |       1  |      8    |

   BITPOOL (8 bits): This unsigned integer indicates the size of the bit
             allocation pool that has been used for encoding the current
             block. The value of the bit-pool field MUST NOT exceed 16
             times the number of subbands for the MONO and DUAL_CHANNEL
             channel modes and 32 times the number of subbands for the
             STEREO and JOINT_STEREO channel modes. The bitpool value
             MAY change from SBC frame to the next. In addition, the
             bitpool value MUST be restricted such that it does not
             result in excess of maximum bit rate, which is 320kb/s for
             mono and 512kb/s for two-channel modes.

  The remaining part of the header consists of CRC_CHECK, optionally
  JOIN bit fields and a RFA.

6.4. Remaining frame

   The remaining part of the frame includes scale factors and audio
   sample data, which are processed by the codec as described in

7. Payload Format Parameters

   This section defines the parameters that MAY be used to configure
   optional features in the SBC payload format over RTP transmission.

   The parameters are defined here as part of the media subtype
   registrations for the SBC. A mapping of the parameters into the
   Session Description Protocol (SDP) [RFC4566] is also provided for
   those applications that use SDP. In control protocols that do not use
   MIME or SDP, the media type parameters must be mapped to the
   appropriate format used with that control protocol.

7.1. SBC Media Type Registration

   [Note to RFC Editor: Please replace all occurrences of RFC XXXX by
   the RFC number assigned to this document]

   This registration is done using the template defined in [RFC4288] and
   following [RFC4855].

   MIME media type name: audio

   MIME subtype name: SBC

   Required parameters: none

   Optional parameters:

   Capabilities: The capabilities of the encoder and decoder are
             described by a parameter string that MUST start with an
             octet written as two hexadecimal digits. This octet is
             called VERSION and MUST be identical to the SYNCWORD that
             will be used in the SBC frames. It is used to distinguish
             different negotiation procedures.
             The interpretation of the following characters depends on
             the value of the VERSION octet. Refer to Section 7.1.1. and
             Section 7.1.2. to find a description.

   Encoding considerations: This media type is framed and contains
             binary data; see Section 4.8 of RFC 4288.

   Security considerations: See Section 9 of RFC XXXX

   Interoperability considerations: none

   Published specification: RFC XXXX

   Applications which use this media type: Audio and video conferencing
             tools, distributed orchestras

   Additional information: none

   Person & email address to contact for further information: Christian

   Intended usage: COMMON

   Restrictions on usage: none

   Author: Christian Hoene, Frans de Bont

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

7.1.1. Capabilities: A2DP modes

   The capabilities of the encoder and decoder MUST start with the
   hexadecimal value of 9C, followed by a comma and four comma-separated
   hexadecimal octets. These four octets called Octet 1, 2, 3, and 4
   share a similar meaning as those defined in Section 4.3.2 of
   [A2DPV10]. However, because sampling frequency and number of channels
   are already given in the SDP parameter "a=rtpmap", bit 0 up to and
   including bit 3 of Octet 1 MUST BE ignored if received. The meaning
   of the bits and the octets are described in the following
   enumeration. The bit numbering follows the network bit order having
   the highest bit first.

   o  Octet 1: Bit 0 (aka 2^7): If one, then the sampling frequency
     16000 Hz is supported (ignored during SDP negotiations but SHOULD
     be set if the clock rate is 16000 and CAN be cleared otherwise).

   o  Octet 1: Bit 1: If one, then the sampling frequency 32000 Hz is
     supported (ignored during SDP negotiations but SHOULD be set if
     the clock rate is 32000 and CAN be cleared otherwise).

   o  Octet 1: Bit 2: If one, then the sampling frequency 44100 Hz is
     supported (ignored during SDP negotiations but SHOULD be set if
     the clock rate is 44100 and CAN be cleared otherwise).

   o  Octet 1: Bit 3: If one, then the sampling frequency 48000 Hz is
     supported (ignored during SDP negotiations but SHOULD be set if
     the clock rate is 48000 and CAN be cleared otherwise).

   o  Octet 1: Bit 4: If one, then the channel mode MONO is supported
     (ignored during SDP negotiations but SHOULD be set if the number
     of channels is one and CAN be cleared otherwise).

   o  Octet 1: Bit 5: If one, then the channel mode DUAL_CHANNEL is
     supported (*).

   o  Octet 1: Bit 6: If one, then the channel mode STEREO is supported

   o  Octet 1: Bit 7 (aka 2^0): If one, then the channel mode
     JOINT_STEREO is supported (*).

   o  Octet 2: Bit 0: If one, the block length can be 4.

   o  Octet 2: Bit 1: If one, the block length can be 8.

   o  Octet 2: Bit 2: If one, the block length can be 12.

   o  Octet 2: Bit 3: If one, the block length can be 16.

   o  Octet 2: Bit 4: If one, the number of subband can be 4.

   o  Octet 2: Bit 5: If one, the number of subband can be 8.

   o  Octet 2: Bit 6: If one, the allocation mode based on signal to
     noise ratio is supported.

   o  Octet 2: Bit 7: If one, the allocation mode based on loudness is

   o  Octet 3: Unsigned integer: The minimal bit-pool value that the
     device supports. MUST be larger or equal than 2 and less or equal
     than the maximal bit-pool value.

   o  Octet 4: Unsigned integer: The maximal bit-pool value that the
     device supports MUST be equal or lower than 250.

   (*) At least one of the bits 5, 6 or 7 of Octet 1 MUST be set if the
      number of channels is set to two in the SDP parameter "a=rtpmap".

7.1.2. Capabilities: other modes

   If the value of the VERSION octet is not equal to a known SYNCWORD
   value, then the capabilities MUST be ignored.

7.2. Mapping to SDP Parameters

   The information carried in the media type specification has a
   specific mapping to fields in the Session Description Protocol (SDP)
   [RFC4566], which is commonly used to describe RTP sessions. When SDP
   is used to specify sessions employing the SBC codec, the mapping is
   as follows:

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

   o  The media subtype ("SBC") goes in SDP "a=rtpmap" as the encoding

   o  The RTP <clock rate> in "a=rtpmap" MUST be set to the selected
     sampling frequency.

   o  The RTP <encoding parameters> in "a=rtpmap" specifies the number
     of audio channels: 2 for stereo material (refer to RFC 4566
     [RFC4566]) and 1 for mono. If one channel is used, the encoding
     parameter can be omitted.

   o  The parameter "capabilities" goes in the SDP "a=fmtp" by the
     capabilities description as described in Section 7.1.

7.2.1. Offer-Answer Model Considerations

   The Bluetooth standard document [AVDTPV12] describes how an A2DP
   source and an A2DP sink negotiate their capabilities. Prior to the
   establishment of the audio stream, one A2DP device can query the
   service capabilities of the other device using the "Get Capabilities
   Procedure". In any case, the coding mode is set using the "Set
   Configuration" procedure. Only after a successful configuration, the
   stream connection can be established.

   In addition to the Bluetooth negotiation procedure, the SDP
   negotiation MUST NOT agree on one single configuration but CAN agree
   that multiple configuration modes, which are identified by different
   payload type values, are supported.

   The following considerations apply when using SDP offer-answer
   procedures [RFC3264] to negotiate the use of SBC payload in RTP:

   o  The "capabilities" parameter is bi-directional, i.e., the
     restricted mode set applies to media both to be received and sent
     by the declaring entity. If the capabilities were supplied in the
     offer, the answerer MUST return either the same mode-set or a
     subset of this mode-set. If no capabilities were supplied in the
     offer, the answerer MAY return capabilities to restrict the
     possible modes. In any case, the capabilities in the answer then
     apply for both offerer and answerer. The offerer MUST NOT send
     frames of a mode that has been removed by the answerer. The
     negotiation is finished if the offerer and the answerer have
     agreed upon explicit capabilities for each payload type number.
     The number of blocks and subbands and the kind of allocation
     method and channel mode MUST haven been negotiated unambiguously.

   o  Any unknown parameter in an offer MUST be ignored by the receiver
     and MUST NOT be included in the answer.

   Below are some example parts of SDP offer-answer exchanges.

   o  Example 1
     Offer: SBC all A2DP modes
              m=audio 54874 RTP/AVP 96
              a=rtpmap:96 SBC/48000/2
              a=fmtp:96 capabilities=9C,17,FF,02,FA
              m=audio 54874 RTP/AVP 97
              a=rtpmap:97 SBC/48000
              a=fmtp:97 capabilities=9C,18,FF,02,FA
              m=audio 54874 RTP/AVP 98
              a=rtpmap:98 SBC/44100/2
              a=fmtp:98 capabilities=9C,27,FF,02,FA
              m=audio 54874 RTP/AVP 99
              a=rtpmap:99 SBC/44100
              a=fmtp:99 capabilities=9C,28,FF,02,FA
              m=audio 54874 RTP/AVP 100
              a=rtpmap:100 SBC/32000/2
              a=fmtp:101 capabilities=9C,47,FF,02,FA
              m=audio 54874 RTP/AVP 102
              a=rtpmap:102 SBC/32000
              a=fmtp:102 capabilities=9C,48,FF,02,FA
              m=audio 54874 RTP/AVP 103
              a=rtpmap:103 SBC/16000/2
              a=fmtp:103 capabilities=9C,87,FF,02,FA
              m=audio 54874 RTP/AVP 104
              a=rtpmap:104 SBC/48000
              a=fmtp:104 capabilities=9C,88,FF,02,FA

      Answer: 48 kHz, JOINT_STEREO, 16 blocks, 8 subbands, LOUDNESS
               m=audio 59452 RTP/AVP 96
               a=rtpmap:96 SBC/48000/2
               a=fmtp:96 capabilities=9C,11,15,02,FA

   o  Example 2
     Offer: The A2DP SBC 48 kHz modes with mono or joint stereo, 8
     subbands, loudness allocation method. In addition an unknown mode
     called AD is offered.
              m=audio 54874 RTP/AVP 96
              a=rtpmap:96 SBC/48000/2
              a=fmtp:96 capabilities=9C,11,F5,02,FA
              m=audio 54874 RTP/AVP 97
              a=rtpmap:97 SBC/48000/1
              a=fmtp:97 capabilities=9C, 18,F5,02,FA
              m=audio 54874 RTP/AVP 98
              a=rtpmap:98 SBC/16000/1
              a=fmtp:98 capabilities=AD

     Answer: both A2DP modes are accepted but the unknown mode AD is
              m=audio 59452 RTP/AVP 96
              a=rtpmap:96 SBC/48000/2
              a=fmtp:96 capabilities=9C,11,F5,02,FA
              m=audio 59452 RTP/AVP 9
              a=rtpmap:97 SBC/48000/1
              a=fmtp:97 capabilities=9C,18,F5,02,FA

7.2.2. Declarative SDP Considerations

   For declarative use of SDP nothing specific is defined for this
   payload format. The configuration given by the SDP MUST be used when
   sending and/or receiving media in the session.

8. Congestion Control

   One Bluetooth links, bandwidth can be reserved and thus the A2DP
   specification does not consider any kind of congestion control.
   However, congestion control is an important issue for any usage in
   non-dedicated networks such as the Internet. Thus, congestion control
   for RTP MUST be used in accordance with [RFC3550] and any appropriate
   profile (for example, [RFC3551]). An additional requirement if best-
   effort service is being used is: users of this payload format MUST
   monitor packet loss to ensure that the packet loss rate is within
   acceptable parameters.

   Reducing the session bandwidth is possible by one or more of the
   following means, which all will have negative impact to the users'
   experience as he can notice a higher latency or a degraded audio
   quality. The selection of the following means depends on current
   usage scenario, the congestion control protocol, and the perceptual
   assessment of the audio transmission and is not subject of this

   1. If the bandwidth and frame rate shall be reduced, the sampling
      rate can be lowered [Boutremans2004,Hoene2005].

   2. If the gross bandwidth and the frame rate shall be reduced, more
      blocks can be put into one SBC frame and more SBC frames can be
      placed in one RTP payload.

   3. If the bandwidth shall be reduced, then the bit-pool value can be
      reduced, so that the frames get smaller or the mono mode can be

   4. If the bandwidth is very low, instead of an ongoing transmission,
      a push-to-talk like service with temporary transmission
      interruptions and a high delay can be applied.

   5. If the packet loss rate is very high, the session shall be
      terminated because the quality of the audio transmission is too
      bad to be useful [Widmer2002].

   Because the SBC encoding can be tuned with many parameters, it is
   especially useful for rate adaptive transport protocols such as DCCP
   [RFC4340] or TCP [RFC4571]. The report [Hoene2009] describes, which
   SBC coding mode gives the best speech and audio quality under known
   bandwidth and time constrains.

9. Packet loss concealment

   In order to cope with packet losses, the SBC decoder SHOULD be
   extended by a packet loss concealment algorithm. The packet loss
   concealment algorithm SHOULD provide a good audio quality in case of
   losses. Otherwise, the congestion control algorithm can not trade off
   well the quality impairment due to packet losses versus the quality
   impairment caused by different encoding modes. It is RECOMMENDED that
   at a least the reserve order replicated pitch periods (RORPP)
   algorithm as defined in [Hoene2009] or any better is used.

   If this requirement is not meet, then the congestion control cannot
   predict the impact of packet loss on the audio quality and thus will
   not be able to control the encoding parameters optimally.

10. Security Considerations

   RTP packets using the payload format defined in this specification
   are subject to the general security considerations discussed in the
   RTP specification [RFC3550] and any appropriate profile (for example,

   As this format transports encoded speech/audio, the main security
   issues include confidentiality, integrity protection, and
   authentication of the speech/audio itself.  The payload format itself
   does not have any built-in security mechanisms.  Any suitable
   external mechanisms, such as SRTP [RFC3711], MAY be used.

   This payload format and the SBC encoding do not exhibit any large
   non-uniformity in the receiver-end computational load and thus are
   unlikely to pose a denial-of-service threat due to the receipt of
   pathological datagrams.

11. IANA Considerations

   It is requested that one new media subtype (audio/SBC) and one
   optional parameter for this media subtype ("capabilities") are
   registered by IANA, see Section 5.1 and Section 5.2.

12. References

12.1. Normative References

   [A2DPV10] Bluetooth SIG, "Advanced Audio Distribution Profile", Audio
             Video WG, adopted specification, revision V1.0, May 22th,

   [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
             Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3264] Rosenberg, J. and Schulzrinne, H., "An Offer/Answer
             Modelwith Session Description Protocol (SDP)", RFC 3264,
             June 2002.

   [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
             Jacobson, "RTP: A Transport Protocol for Real-Time
             Applications", STD 64, RFC 3550, July 2003.

   [RFC3551] Schulzrinne, H. and Casner, S., "RTP Profile for Audio and
             Video Conferences with Minimal Control", STD 65, RFC 3551,
             July 2003.

   [RFC4288] Freed, N. and Klensin, J., "Media Type Specifications and
             Registration Procedures", BCP 13, RFC 4288, December 2005.

   [RFC4566] Handley, M., Jacobson, V., and Perkins, C., "SDP: Session
             Description Protocol", RFC 4566, July 2006.

   [RFC4855] Casner, S., "Media Type Registration of RTP Payload
             Formats", RFC 4855, February 2007.

12.2. Informative References

   [AVDTPV12] Bluetooth SIG, "Audio/Video Distribution Transport
             Protocol Specification", Audio Video WG, adopted
             specification, revision V12, April 16th, 2007.

   [Bon1995] de Bont, F., Groenewegen, M., and Oomen, W., "A High
             Quality Audio-Coding System at 128 kb/s", 98th AES
             Convention, February 25 - 28, 1995.

   [Boutremans2004] Boutremans, C., Le Boudec J.-Y., and Widmer, J.,
             "End-to-end congestion control for tcp-friendly flows with
             variable packet size", ACM Computer Communication Review,
             Vol. 31, No. 2, pp. 137-151, 2004.

   [Pilati2008] Pilati, L., Zadissa, M., "Enhancements to the SBC CODEC
             for Voice Communication in Mobile Devices", AES Convention
             124, No. 7347, May 2008.

   [Hoene2009] Hoene, C., Hyder, M.. "Considering bluetooth's subband
             codec (SBC) for wideband speech and audio on the internet".
             Technical Report WSI-2009-3, Universitaet Tuebingen - WSI,
             72076 Tuebingen, Germany, October 2009.

   [GAVDPV12] Bluetooth SIG, "Generic Audio/Video Distribution Profile",
             Audio Video WG, adopted specification, revision V12, April
             16th, 2007.

   [Gurevich2004] Gurevich, M., Chafe, C., Leslie, G., and Tyan, S.,
             "Simulation of Networked Ensemble Performance with Varying
             Time Delays: Characterization of Ensemble Accuracy",
             Proceedings of the 2004 International Computer Music
             Conference, Miami, USA, 2004.

   [Hoene2005] Hoene, C., and Karl, H., and Wolisz, A., "A perceptual
             quality model intended for adaptive VoIP applications",
             International Journal of Communication Systems, Wiley,
             August 2005.

   [ITUG107] ITU-T G.107, "The E-model, a computational model for use in
             transmission planning", ITU-T Recommendation G.107, May

   [Rault1989] Rault, J., Dehery, Y., Roudaut, J., Bruekers, A., and
             Veldhuis, R., "Digital transmission system using subband
             coding of a digital signal", Publication number: EP0400755

   [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
             Norrman, "The Secure Real-time Transport Protocol (SRTP)",
             RFC 3711, March 2004.

   [RFC4340] Kohler, E., Handley, M., and Floyd, S., "Datagram
             Congestion Control Protocol (DCCP)", RFC 4340, March 2006.

   [RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP)
             and RTP Control Protocol (RTCP) Packets over Connection-
             Oriented Transport", RFC4571, July 2006.

   [Widmer2002] Widmer, J., Mauve, M., and Damm, J., "Probabilistic
             congestion control for non-adaptable flows", In 12th
             International Workshop on Network and Operating Systems
             Support for Digital Audio and Video (NOSSDAV), Miami, FL,
             USA, May 2002.

13. Acknowledgments

   Funding for this draft has been provided by the University of
   Tuebingen within the "Projektfoerderung fuer

   This document was prepared using

Authors' Addresses

   Christian Hoene
   University of Tuebingen
   Sand 13
   72076 Tuebingen

   Phone: +49 7071 29 70532

   Frans de Bont
   Philips Electronics
   High Tech Campus 5
   5656 AE Eindhoven

   Phone: +31 40 2740234