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RTP Payload Format for G.719

The information below is for an old version of the document that is already published as an RFC.
Document Type
This is an older version of an Internet-Draft that was ultimately published as RFC 5404.
Authors Magnus Westerlund , Ingemar Johansson
Last updated 2020-01-21 (Latest revision 2008-11-17)
Replaces draft-westerlund-avt-rtp-g719
RFC stream Internet Engineering Task Force (IETF)
Intended RFC status Proposed Standard
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IESG IESG state Became RFC 5404 (Proposed Standard)
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Responsible AD Cullen Fluffy Jennings
Send notices to (None)
Network Working Group                                      M. Westerlund
Internet-Draft                                              I. Johansson
Intended status: Standards Track                             Ericsson AB
Expires: May 21, 2009                                       Nov 17, 2008

                      RTP Payload format for G.719

Status of this Memo

   By submitting this Internet-Draft, each author represents that any
   applicable patent or other IPR claims of which he or she is aware
   have been or will be disclosed, and any of which he or she becomes
   aware will be disclosed, in accordance with Section 6 of BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
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   The list of current Internet-Drafts can be accessed at

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   This Internet-Draft will expire on May 21, 2009.


   This document specifies the payload format for packetization of the
   G.719 full-band codec encoded audio signals into the Real-time
   Transport Protocol (RTP).  The payload format supports transmission
   of multiple channels, multiple frames per payload, and interleaving.

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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Definitions and Conventions  . . . . . . . . . . . . . . . . .  3
   3.  G.719 Description  . . . . . . . . . . . . . . . . . . . . . .  3
   4.  Payload format Capabilities  . . . . . . . . . . . . . . . . .  4
     4.1.  Multi-rate Encoding and Rate Adaptation  . . . . . . . . .  4
     4.2.  Support for Multi-Channel Sessions . . . . . . . . . . . .  5
     4.3.  Robustness against Packet Loss . . . . . . . . . . . . . .  5
       4.3.1.  Use of Forward Error Correction (FEC)  . . . . . . . .  5
       4.3.2.  Use of Frame Interleaving  . . . . . . . . . . . . . .  6
   5.  Payload format . . . . . . . . . . . . . . . . . . . . . . . .  7
     5.1.  RTP Header Usage . . . . . . . . . . . . . . . . . . . . .  8
     5.2.  Payload Structure  . . . . . . . . . . . . . . . . . . . .  8
       5.2.1.  Basic ToC element  . . . . . . . . . . . . . . . . . .  9
     5.3.  Basic mode . . . . . . . . . . . . . . . . . . . . . . . . 10
     5.4.  Interleaved mode . . . . . . . . . . . . . . . . . . . . . 10
     5.5.  Audio Data . . . . . . . . . . . . . . . . . . . . . . . . 11
     5.6.  Implementation Considerations  . . . . . . . . . . . . . . 12
       5.6.1.  Receiving Redundant Frames . . . . . . . . . . . . . . 12
       5.6.2.  Interleaving . . . . . . . . . . . . . . . . . . . . . 12
       5.6.3.  Decoding Validation  . . . . . . . . . . . . . . . . . 13
   6.  Payload Examples . . . . . . . . . . . . . . . . . . . . . . . 13
     6.1.  3 mono frames with 2 different bitrates  . . . . . . . . . 14
     6.2.  2 stereo frame-blocks of the same bitrate  . . . . . . . . 14
     6.3.  4 mono frames interleaved  . . . . . . . . . . . . . . . . 15
   7.  Payload Format Parameters  . . . . . . . . . . . . . . . . . . 16
     7.1.  Media Type Definition  . . . . . . . . . . . . . . . . . . 16
     7.2.  Mapping to SDP . . . . . . . . . . . . . . . . . . . . . . 19
       7.2.1.  Offer/Answer Considerations  . . . . . . . . . . . . . 20
       7.2.2.  Declarative SDP Considerations . . . . . . . . . . . . 23
   8.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 23
   9.  Congestion Control . . . . . . . . . . . . . . . . . . . . . . 23
   10. Security Considerations  . . . . . . . . . . . . . . . . . . . 24
     10.1. Confidentiality  . . . . . . . . . . . . . . . . . . . . . 24
     10.2. Authentication and Integrity . . . . . . . . . . . . . . . 25
   11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 25
   12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 25
     12.1. Informative References . . . . . . . . . . . . . . . . . . 25
     12.2. Normative References . . . . . . . . . . . . . . . . . . . 26
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 27
   Intellectual Property and Copyright Statements . . . . . . . . . . 28

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1.  Introduction

   This document specifies the payload format for packetization of the
   G.719 full-band (FB) codec encoded audio signals into the Real-time
   Transport Protocol (RTP) [RFC3550].  The payload format supports
   transmission of multiple channels, multiple frames per payload,
   packet loss robustness methods using redundancy or interleaving.

   This document starts with conventions, a brief description of the
   codec, and the payload formats capabilities.  The payload format is
   specified in Section 5.  Examples can be found in Section 6.  The
   media type and its mappings to SDP, usage in SDP offer/answer is then
   specified.  The document ends with considerations around congestion
   control and security.

2.  Definitions and Conventions

   The term "frame-block" is used in this document to describe the time-
   synchronized set of audio frames in a multi-channel audio session.
   In particular, in an N-channel session, a frame-block will contain N
   audio frames, one from each of the channels, and all N speech frames
   represents exactly the same time period.

   This document contains depictions of bit fields.  The most
   significant bit is always leftmost in the figure on each row and have
   the lowest enumeration.  For fields that are depicted over multiple
   rows the upper row is more significant than the next.

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

3.  G.719 Description

   The ITU-T G.719 full-band codec is a transform coder based on
   Modulated Lapped Transform (MLT).  G.719 is a low complexity full
   bandwidth codec for conversational speech and audio coding.  The
   encoder input and decoder output are sampled at 48 kHz.  The codec
   enables full bandwidth, from 20 Hz to 20 kHz, encoding of speech,
   music and general audio content at rates from 32 kbit/s up to 128
   kbit/s.  The codec operates on 20ms frames and has an algorithmic
   delay of 40 ms.

   The codec provides excellent quality for speech, music and other
   types of audio.  Some of the applications for which this coder is
   suitable are:

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   o  Real-time communications such as video conferencing and telephony.

   o  Streaming audio

   o  Archival and messaging

   The encoding and decoding algorithm can change the bit rate at any
   20ms frame boundary.  The encoder receives the audio sampled at
   48kHz.  The support of other sampling rates is possible by re-
   sampling the input signal to the codec's sampling rate, i.e. 48kHz,
   however, this functionality is not part of the standard.

   The encoding is performed on equally sized frames.  For each frame,
   the encoder decides between two encoding modes, a transient mode and
   a stationary mode.  The decision is based on statistics derived from
   the input signal.  The stationary mode uses a long MLT that leads to
   a spectrum of 960 coefficients while the transient encoding mode uses
   a short MLT (higher time resolution transform) which results in 4
   spectra (4 x 240 = 960 coefficients).  The encoding of the spectrum
   is done in two steps.  First, the spectral envelope is computed,
   quantized and Huffman encoded.  The envelope is computed on a non-
   uniform frequency subdivision.  From the coded spectral envelope, a
   weighted spectral envelope is derived and is used for bit-allocation,
   this process is also repeated at the decoder, thus only the spectral
   envelope is transmitted.  The output of the bit-allocation is used in
   order to quantize the spectra.  In addition, for stationary frames
   the encoder estimates the amount of noise level.  The decoder applies
   the reverse operation upon reception of the bit stream.  The non-
   coded coefficients (i.e. no bits allocated) are replaced by entries
   of a noise codebook which is built based on the decoded coefficients.

4.  Payload format Capabilities

   This payload format have a number of capabilities and this section
   discuss them in some detail.

4.1.  Multi-rate Encoding and Rate Adaptation

   G.719 supports multi-rate encoding capability that enables on a per
   frame basis variation of the encoding rate.  This enables support for
   bit-rate adaptation and congestion control.  The possibility to
   aggregate multiple audio frames into a single RTP payload is another
   dimension of adaptation.  The RTP and payload format overhead can
   thus be reduced by the aggregation at the cost of increased delay and
   reduced packet-loss robustness.

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4.2.  Support for Multi-Channel Sessions

   The RTP payload format defined in this document supports multi-
   channel audio content (e.g. stereophonic or surround audio sessions).
   Although the G.719 codec itself does not support encoding of multi-
   channel audio content into a single bit stream, it can be used to
   separately encode and decode each of the individual channels.  To
   transport (or store) the separately encoded multi-channel content,
   the audio frames for all channels that are framed and encoded for the
   same 20 ms period are logically collected in a "frame-block".

   At the session setup, out-of-band signaling must be used to indicate
   the number of channels in the payload type.  The order of the audio
   frames within the frame-block depends on the number of the channels
   and follows the definition in Section 4.1 of the RTP/AVP Profile
   [RFC3551].  When using SDP for signaling, the number of channels is
   specified in the rtpmap attribute.

4.3.  Robustness against Packet Loss

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

4.3.1.  Use of Forward Error Correction (FEC)

   Generic forward error correction within RTP is defined, for example,
   in RFC 5109 [RFC5109].  Audio redundancy coding is defined in RFC
   2198 [RFC2198].  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 RFCs 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
   optional G.719 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 can be used to group the frames to be
   sent in each payload.  However, irregular or non-consecutive patterns
   are also possible by inserting NO_DATA frames between primary and
   redundant transmissions.  Figure 1 below shows an example.

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

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

              Figure 1: An example of redundant transmission

   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 really require signaling at the
   session setup.  However, signalling has been defined to allow for the
   sender to voluntarily bounding the buffering and delay requirements.
   If nothing is signalled the use of this mechanism is allowed and
   unbounded.  For a certain timestamp, the receiver may receive
   multiple copies of a frame containing encoded audio data, even at
   different encoding rates.  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 G.719 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) [RFC3550] receiver reports.  The
   sender is also responsible for avoiding congestion, which may be
   exacerbated by redundancy (see Section 9 for more details).

4.3.2.  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,

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   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

   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.

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

              [ p(n)   ]
     [ p(n+1) ]                 [ p(n+1) ]
                       [ p(n+2) ]                 [ p(n+2) ]
                                         [ p(n+3) ]
                                                           [ 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 de-interleaving
   processes.  The de-interleaving 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 de-interleaving 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.

5.  Payload format

   The main purpose of the payload design for G.719 is to maximize the
   potential of the codec to its fullest degree with an as minimal
   overhead as possible.  In the design both basic and interleaved modes

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   have been included as the codec is suitable both for conversational
   and other low delay applications as well as streaming, where more
   delay is acceptable.

   The main structural difference between the basic and interleaved
   modes is the extension of the table of content entries with frame
   displacement fields 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 payload format also supports the usage of G.719 for carrying
   multi-channel content using one discrete encoder per channel all
   using the same bit-rate.  In this case a complete frame-block with
   data from all channels are included in the RTP payload.  The data is
   the concatenation of all the encoded audio frames in the order
   specified for that number of included channels.  Also interleaving is
   done on complete frame-blocks rather than individual audio frames.

5.1.  RTP Header Usage

   The RTP timestamp corresponds to the sampling instant of the first
   sample encoded for the first frame-block in the packet.  The
   timestamp clock frequency SHALL be 48000 Hz.  The timestamp is also
   used to recover the correct decoding order of the frame-blocks.

   The RTP header marker bit (M) SHALL be set to 1 whenever the first
   frame-block carried in the packet is the first frame-block in a
   talkspurt (see definition of the talkspurt in section 4.1 of
   [RFC3551]).  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 profiles in use
   currently mandates binding the payload type dynamically for this
   payload format.  This is basically necessary due to that the payload
   type expresses the configuration of the payload itself, i.e. basic or
   interleaved mode and the number of channels carried.

   The remaining RTP header fields are used as specified in RFC 3550

5.2.  Payload Structure

   The payload consists of one or more table of contents (ToC) entries
   followed by the audio data corresponding to the ToC entries.  The
   following sections describe both the basic mode and the interleaved

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   mode.  Each ToC entry MUST be padded to a byte boundary to ensure
   octet alignment.  The rules regarding maximum payload size given in
   Section 3.2 of [I-D.ietf-tsvwg-udp-guidelines] SHOULD be followed.

5.2.1.  Basic ToC element

   All the different formats and modes in this draft use a common basic
   ToC which may be extended in the different options described below.

    0 1 2 3 4 5 6 7
   |F|    L    |R|R|

                        Figure 3: Basic TOC element

   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.

   L (5 bits):  A field that gives the frame length of each individual
      frame within the frame-block.

        L          length(bytes)
        0           0 NO_DATA
        1-7         N/A (reserved)
        8-22        80+10*(L-8)
       23-27        240+20*(L-23)
       28-31        N/A (reserved)

                Figure 4: How to map L values to frame lengths

      L=0 (NO_DATA) is used to indicate an empty frame, this is useful
      if frames are missing e.g at re-packetization or to insert gaps
      when sending redundant frames together with primary frames in the
      same payload.
      The value range [1..7] and [28..31] inclusive is reserved for
      future use in this draft version, if these values occur in a ToC
      the entire packet SHOULD be treated as invalid and discarded.
      A few examples are given below where the frame size and the
      corresponding codec bitrate is computed based on the value L.

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         L    Bytes    Codec Bitrate(kbps)
         8      80        32
         9      90        36
        10     100        40
        12     120        48
        16     160        64
        22     220        88
        23     240        96
        25     280       112
        27     320       128

        Figure 5: Examples of L values and corresponding frame lengths

      This encoding yields a granularity of 4kbps between 32 and 88kbps
      and a granularity of 8kbps between 88 and 128kbps with a defined
      range of 32-128kbps for the codec data.

   R (2bits):  Reserved bits.  SHALL be set to 0 on sending and SHALL be
      ignored on reception.

5.3.  Basic mode

   The basic ToC element Figure 3 is followed by a one octet field for
   the number of frame-blocks (#frames) to form the ToC entry.  The
   frame-blocks field tells how many frame-blocks of the same length the
   ToC entry relates to.

    0 1 2 3 4 5 6 7
   |    #frames    |

                  Figure 6: Number of frame-blocks field

5.4.  Interleaved mode

   The basic ToC is followed by a one octet field for the number of
   frame-blocks (#frames) and then the DIS fields to form a ToC entry in
   interleaved mode.  The frame-blocks field tells how many frame-blocks
   of the same length the ToC relates to.  The DIS fields, one for each
   frame-block indicated by the #frames field, express the interleaving
   distance between audio frames carried in the payload.  If necessary
   to achieve octet alignment, a 4-bit padding is added.

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   |    #frames    | DIS1  |  ...  | DISi  |  ...  | DISn  | Padd  |

            Figure 7: Number of frame-block + interleave fields

   DIS1...DISn (4 bits):  A list of n (n=#frames) displacement fields
      indicating the displacement of the i:th (i=1..n) audio frame-block
      relative to the preceding frame-block in the payload, in units of
      20 ms long audio frame-blocks).  The four-bit unsigned integer
      displacement values may be between 0 and 15 indicating the number
      of audio frame-blocks 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-block's location in the decoding order is uniquely defined
      by the RTP timestamp.  Note 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.

   Padd (4 bits):  To ensure octet alignment, four padding bits SHALL be
      included at the end of the ToC entry in case there is an odd
      number of frame-blocks in the group referenced by this ToC 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.

5.5.  Audio Data

   The audio data part follows the table of contents.  All the octets
   comprising an audio frame SHALL be appended to the payload as a unit.
   For each frame-block the audio frames are concatenated in order
   indicated by table in Section 4.1 of [RFC3551] for the number of
   channels configured for the payload type in use.  So the first
   channel (left most) indicated comes first followed by the next
   channel.  The audio frame-blocks are packetized in increasing
   timestamp order within each group of frame-blocks (per ToC entry),
   i.e. oldest frame-block first.  The groups of frame-blocks are
   packetized in the same order as their corresponding ToC entries.

   The audio frames are specified in ITU recommendation [ITU-T-G719].

   The G.719 bit stream is split into a sequence of octets and
   transmitted in order from the left most (most significant-MSB) bit to
   the right most (least significant -LSB) bit.

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5.6.  Implementation Considerations

   An application implementing this payload format MUST understand all
   the payload parameters specified in this specification.  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 when the communicating entities support this version of
   the specification.

   Basic mode SHALL be implemented and the interleaved mode SHOULD be
   implemented.  The implementation burden of both is rather small, and
   supporting both ensures interoperability.  However, interleaving is
   not mandated as it has limited applicability for conversational
   application that requires tight delay boundaries.

5.6.1.  Receiving Redundant Frames

   The reception of redundant audio frames, i.e. more than one audio
   frame from the same source for the same time slot, MUST be supported
   by the implementation.  In the case that the receiver gets multiple
   audio frames in different bit-rates for the same time slot it is
   RECOMMENDED that the receiver keeps the one with the highest bit-

5.6.2.  Interleaving

   The use of interleaving requires further considerations.  As
   presented in the example in Section 4.3.2, a given interleaving
   pattern requires a certain amount of the de-interleaving 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 320 bytes per frame used by the highest bit-rate
   rate of G.719.

   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.  Additional information is needed 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 the interleaving pattern, it
   can calculate this value.  In certain cases (for example, if joining
   a multicast session with interleaving mid-session), a receiver may

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   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 G.719 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

   In order to handle this type of startup efficiently, decoding can
   start provided that:

   1.  There are at least two consecutive frames available.

   2.  More than or equal to half the frames are available in the time
       period from where decoding was planned to start and the most
       forward received decoding.

   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.  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 they
   receiving content for which they do not have the key, it is
   recommended that interleaving patterns do not overlap key changes.

5.6.3.  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.

6.  Payload Examples

   A few examples to highlight the payload format

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6.1.  3 mono frames with 2 different bitrates

   The first example is a payload consisting of 3 mono frames where the
   2 first frames correspond to a bitrate of 32kbps (80byte/frame) and
   the last is 48kbps (120byte/frame).

      The first 32 bits are ToC fields.
      Bit 0 is '1' as another ToC field follow.
      Bits 1..5 is 01000 = 80bytes/frame
      Bits 8..15 is 00000010 = 2 frame-blocks with 80bytes/frame
      Bit 16 is '0', no more ToC follows
      Bits 17..21 is 01100 = 120 bytes/frame
      Bits 24..31 = 00000001 = 1 frame-block with 120bytes/frame

       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|0 1 0 0 0|0 0|0 0 0 0 0 0 1 0|0|0 1 1 0 0|0 0|0 0 0 0 0 0 0 1|
      |d(0)   frame 1                                                 |
      .                                                               .
      |                                                         d(639)|
      |d(0)   frame 2                                                 |
      .                                                               .
      |                                                         d(639)|
      |d(0)   frame 3                                                 |
      .                                                               .
      |                                                         d(959)|

6.2.  2 stereo frame-blocks of the same bitrate

   A payload consisting of 2 stereo frames corresponding to a bitrate of
   32kbps (80byte/frame) per channel.  The receiver calculates the
   number of frames in the audio block by multiplying the value of the
   channels parameter (2) with the #frames field value (2) to derive
   that there are 4 audio frames in the payload.

      The first 16 bits is the ToC field.
      Bit 0 is '0' as no ToC field follow.
      Bits 1..5 is 01000 = 80bytes/frame
      Bits 8..15 is 00000010 = 2 frame-blocks with 80bytes/frame

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       0                   1                   2                   3
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
      |0|0 1 0 0 0|0 0|0 0 0 0 0 0 1 0| d(0) frame 1 left ch.         |
      .                                                               .
      |                         d(639)| d(0) frame 1 right ch.        |
      .                                                               .
      |                         d(639)| d(0) frame 2 left ch.         |
      .                                                               .
      |                         d(639)| d(0) frame 2 right ch.        |
      |                         d(639)|

6.3.  4 mono frames interleaved

   A payload consisting of 4 mono frames corresponding to a bitrate of
   32kbps (80byte/frame) interleaved.  A pattern of interleaving for
   constant delay when aggregating 4 frames is used in the below
   example.  The actual packet illustrated is packet n, while the
   previous and following packets frame-block content is shown to
   illustrate the pattern.

      Packet n-3:  1,  6, 11, 16
      Packet n-2:  5, 10, 15, 20
      Packet n-1:  9, 14, 19, 24
      Packet   n: 13, 18, 23, 28
      Packet n+1: 17, 22, 27, 32
      Packet n+2: 21, 26, 31, 36

      The first 16 bits is the ToC field.
      Bit 0 is '0' as there are no ToC field following.
      Bits 1..5 is 01000 = 80bytes/frame
      Bits 8..15 is 00000100 = 4 frame-blocks with 80bytes/frame
      Bits 16..19 is 0000 = DIS1 (0)
      Bits 20..23 is 0100 = DIS2 (4)
      Bits 24..27 is 0100 = DIS3 (4)
      Bits 28..31 is 0100 = DIS4 (4)

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       0                   1                   2                   3
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
      |0|0 1 0 0 0|0 0|0 0 0 0 0 1 0 0|0 0 0 0|0 1 0 0|0 1 0 0|0 1 0 0|
      | d(0) frame 13                                                 |
      .                                                               .
      |                                                         d(639)|
      | d(0) frame 18                                                 |
      .                                                               .
      |                                                         d(639)|
      | d(0) frame 23                                                 |
      .                                                               .
      |                                                         d(639)|
      | d(0) frame 28                                                 |
      .                                                               .
      |                                                         d(639)|

7.  Payload Format Parameters

   This RTP payload format is identified using the media type audio/g719
   which is registered in accordance with [RFC4855] and using the
   template of [RFC4288].

7.1.  Media Type Definition

   The media type for the G.719 codec is allocated from the IETF tree
   since G.719 is a has the potential to become a widely used audio
   codec in general VoIP, teleconferencing and streaming applications.
   This media type registration covers real-time transfer via RTP.

   Note, any unspecified parameter MUST be ignored by the receiver to
   ensure that additional parameters can be added in any future revision
   of this specification.

   Type name: audio

   Subtype name: g719

   Required parameters: none

   Optional parameters:

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   interleaving:  Indicates that interleaved mode SHALL be used for the
      payload.  The parameter specifies the number of frame-block slots
      available in a de-interleaving buffer (including the frame that is
      ready to be consumed).  Its value is equal to one plus the maximum
      number of frames that can 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 milliseconds that is
      needed to avoid underrun in the de-interleaving buffer before
      starting decoding, i.e., the difference in RTP timestamp ticks
      between the earliest and latest audio frame present in the de-
      interleaving buffer expressed in milliseconds.  The value is a
      stream property and provided per source.  The allowed values are 0
      to the largest value expressible by a unsigned 16 bit integer
      (65535).  Please note that the in practice largest value that can
      be used is equal to the declared size of the interleaving buffer
      of the receiver.  If the value for some reason is larger than the
      receiver buffer declared by or for the receiver this value
      defaults to the size of the receiver buffer.  For sources for
      which this value hasn't been provided the value defaults to the
      size of the receiver buffer.  The format is comma separated list
      of SSRC ":" delay in ms pairs which in ABNF [RFC5234] is expressed

         int-delay = "int-delay:" source-delay *("," source-delay)

         source-delay = SSRC ":" delay-value

         SSRC = 1*8HEXDIG ; The 32-bit SSRC encoded in hex format

         delay-value = 1*5DIGIT ; The delay value in milliseconds

         Example: int-delay=ABCD1234:1000,4321DCB:640

         NOTE: No white space allowed in the parameter before the end of
         all the value pairs

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

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   channels:  The number of audio channels.  The possible values (1-6)
      and their respective channel order is specified in Section 4.1 in
      [RFC3551].  If omitted, it has the default value of 1.

   CBR:  Constant Bit Rate (CBR), indicates the exact codec-bitrate in
      bits per second (not including the overhead from packetization,
      RTP header or lower layers) that the codec MUST use.  CBR is to be
      used when dynamic rate cannot be supported (one case is e.g
      gateway to H.320).  CBR is mostly used for gateways to circuit
      switch networks.  Therefore the CBR rate is the rate not including
      any FEC as specified in Section 4.3.1.  If FEC is to be used the
      b= parameter MUST be used to allow the extra bit rate needed to
      send the redundant information.  It is RECOMMENDED that this
      parameter is only used when necessary to establish a working
      communication.  The usage of this parameter have implications on
      congestion control that needs to be considered, see Section 9.

   ptime:  see [RFC4566].

   maxptime:  see [RFC4566].

   Encoding considerations:

      This media type is framed and binary, see section 4.8 in RFC4288

   Security considerations:

      See Section 10 of RFC XXXX.

   Interoperability considerations:

      The support of the Interleaving mode is not mandatory and needs to
      be negotiated.  See Section 7.2 for how to that for SDP based

   Published specification:

      RFC XXXX

   Applications that use this media type:

      Real-time audio applications like voice over IP and
      teleconference, and multi-media streaming.

   Additional information: none

   Person & email address to contact for further information:

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      Payload format: IngemarJohansson

   Intended usage: COMMON

   Restrictions on usage:

      This media type depends on RTP framing, and hence is only defined
      for transfer via RTP [RFC3550].  Transport within other framing
      protocols is not defined at this time.


      Ingemar Johansson <>

      Magnus Westerlund <>

   Change controller:

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

   Additional Information:

      File storage of G.719 encoded audio in ISO base media file format
      is specified in Annex A of [ITU-T-G719].  Thus media file formats
      such as MP4 (audio/mp4 or video/mp4) [RFC4337] and 3GP (audio/3GPP
      and video/3GPP) [RFC3839] can contain G.719 encoded audio.

7.2.  Mapping to SDP

   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 G.719 codec, the mapping is
   as follows:

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

   o  The media subtype (payload format name) goes in SDP "a=rtpmap" as
      the encoding name.  The RTP clock rate in "a=rtpmap" MUST be
      48000, and the encoding parameter "channels" (Section 7.1) MUST
      either be explicitly set to N or omitted, implying a default value
      of 1.  The values of N that are allowed are specified in Section
      4.1 in [RFC3551].

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

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

7.2.1.  Offer/Answer Considerations

   The following considerations apply when using SDP Offer-Answer
   procedures to negotiate the use of G.719 payload in RTP:

   o  Each combination of the RTP payload transport format configuration
      parameters (interleaving, and channels) is unique in its bit-
      pattern and not compatible with any other combination.  When
      creating an offer in an application desiring to use the more
      advanced features (interleaving, or more than one channel), the
      offerer is RECOMMENDED to also offer a payload type containing
      only the configuration with a single channel.  If multiple
      configurations are of interest to the application, they may all be
      offered; however, care should be taken not to offer too many
      payload types.  An SDP answerer MUST include, in the SDP answer
      for a payload type, the following parameters unmodified from the
      SDP offer (unless it removes the payload type): "interleaving";
      and "channels".  However, the value of the Interleaving parameter
      MAY be changed.  The SDP offerer and answerer MUST generate G.719
      packets as described by these parameters.

   o  The "interleaving" and "int-delay" parameter's values have a
      specific relationship that needs to be considered.  It also
      depends on the directionality of the streams and their delivery
      method.  The high level explanation that can be understood from
      the definition is that the value of "interleaving" declares the
      size of the receiver buffer, while int-delay is a stream property
      provided by the sender to inform how much buffer space it in
      practice is using for the stream it sends.

      *  For media streams which is sent over multicast the value of
         "interleaving" SHALL NOT be changed by the answerer.  It shall
         either be accepted or the payload type deleted.  The value of
         the "int-delay" parameter is a stream property and provided by
         the offer/answer agent that intends to send media with this
         payload type, and for each stream coming from that agent (one
         or more).  The value MUST be between 0 and what corresponds to
         the buffer size declared by the value of the "interleaving"

      *  For unicast streams which the offerer declares as send-only the
         value of the "interleaving" parameter is the size that the
         answerer is RECOMMENDED to use by the offerer.  The answerer
         MAY change it to any allowed value.  The int-delay parameter

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         value will be the one the offerer intends to use unless the
         answerer reduce the value of the interleaving parameter below
         what is needed for that int-delay value.  If the interleaving
         value in the answer is smaller than the offer's int-delay, the
         int-delay value is per default reduced to be corresponding to
         the interleaving value.  If the offerer is not satisfied with
         this he will need to perform another round of offer/answer.  As
         the answerer will not send any media it doesn't include any
         int-delay in the answer.

      *  For unicast streams which the offerer declares as recvonly the
         value of interleaving in the offer will be the offerer's size
         of the interleaving buffer.  The answerer indicate its
         preferred size of the interleaving buffer for any future round
         of offer/answer.  The offerer will not provide any int-delay
         parameter as it is not sending any media.  The answerer is
         recommended in its answer include a int-delay parameter to
         declare what the property is for the stream it is going to
         send.  As it already know the receivers interleaving buffer
         size, there should be no issue with providing a value that is
         between 0 and corresponding to a full de-interleaving buffer.

      *  For unicast streams which the offer declares as sendrecv
         streams the value of the interleaving parameter in the offer
         will be offerer's size of the interleaving buffer.  The
         answerer will in the answer indicate the size of its actual
         interleaving buffer.  It is recommended that this value is as
         least as big as the offer's.  The offerer is recommended to
         include a int-delay parameter that is selected based on that
         the answerer has at least as much interleaving space as the
         offerer unless nothing else is known.  As the offerer's
         interleaving buffer size is not yet known this may fail, in
         which cases the default rule is to downgrade the value of the
         int-delay to correspond to the full size of the answerer's
         interleaving buffer.  If the offerer isn't satisfied with this
         it will need to initiate another round of offer/answer.  The
         answerer is recommended in its answer include a int-delay
         parameter to declare what the property is for the stream(s) it
         is going to send.  As it already know the receivers
         interleaving buffer size, there should be no issue with
         providing a value that is between 0 and corresponding to a full
         de-interleaving buffer.

   o  In most cases, the parameters "maxptime" and "ptime" will not
      affect interoperability; however, the setting of the parameters
      can affect the performance of the application.  The SDP offer-
      answer handling of the "ptime" parameter is described in
      [RFC3264].  The "maxptime" parameter MUST be handled in the same

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   o  The parameter "max-red" is a stream property parameter.  For
      sendonly or sendrecv unicast media streams, the parameter declares
      the limitation on redundancy that the stream sender will use.  For
      recvonly streams, it indicates the desired value for the stream
      sent to the receiver.  The answerer MAY change the value, but is
      RECOMMENDED to use the same limitation as the offer declares.  In
      the case of multicast, the offerer MAY declare a limitation; this
      SHALL be answered using the same value.  A media sender using this
      payload format is RECOMMENDED to always include the "max-red"
      parameter.  This information is likely to simplify the media
      stream handling in the receiver.  This is especially true if no
      redundancy will be used, in which case "max-red" is set to 0.

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

   o  The b= SDP parameter SHOULD be used to negotiate the maximum
      bandwidth to be used for the audio stream.  The offerer may offer
      a maximum rate and the answer may contain a lower rate.  If no b=
      parameter is present in the offer or answer it implies a rate up
      to 128kbps

   o  The parameter "CBR" is a receiver capability, i.e. only receivers
      that really requires constant bit-rate should use it.  Usage of
      this parameter have negative impact on the possibility to perform
      congestion control, see Section 9.  For recvonly and sendrecv
      streams, it indicates the desired constant bit rate that the
      receiver wants to accept.  A sender MUST be able to send constant
      bit rate stream since it is a subset of the variable bit rate
      capability.  If the offer includes this parameter the answerer
      MUST send G.719 audio at the constant bit rate if it is within the
      allowed session bit rate (b= parameter).  If the answerer can not
      support the stated CBR this payload type must be refused in the
      answer.  The answerer SHOULD only include this parameter if it
      self requires to receive at a constant bit rate, even if the offer
      did not include the CBR parameter.  In this case, the offerer
      SHALL send at the constant bit rate but SHALL be able to accept
      media at variable bit rate.  An answerer is RECOMMEND to use the
      same CBR rate as in the offer, as symmetric usage is more likely
      to work.  If both sides requires a particular CBR rate there is
      the possibility of communication failure when one or both sides
      can't transmit the requested rate.  In this case the agent
      detecting this issue will have to perform a second round of offer/
      answer to try to find another working configuration or end the
      established session.  In case the offer contained a CBR parameter
      but the answer does not, then the offerer is free to transmit at
      any rate to the answerer, but the answerer is restricted to the

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      declared rate.

7.2.2.  Declarative SDP Considerations

   In declarative usage, like SDP in RTSP [RFC2326] or SAP [RFC2974],
   the parameters SHALL be interpreted as follows:

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

   o  It might not be possible to know the SSRC values that are going to
      be used by the sources at the time of sending the SDP.  This is
      not a major issues as the size of the interleaving buffer can be
      tailored towards the values actually going to be used.  Thus
      ensuring that the default values for int-delay is not resulting in
      to much extra buffering.

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

   o  The parameter "CBR" if included applies to all RTP streams using
      that payload type for which a particular CBR rate is declared.
      Usage of this parameter have negative impact on the possibility to
      perform congestion control, see Section 9.

8.  IANA Considerations

   One media type (audio/g719) has been defined and needs registration
   in the media types registry; see Section 7.1.

9.  Congestion Control

   The general congestion control considerations for transporting RTP
   data apply; see RTP [RFC3550] and any applicable RTP profile like AVP
   [RFC3551].  However, the multi-rate capability of G.719 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 encoding bit-rate.

   The number of frames encapsulated in each RTP payload highly
   influences the overall bandwidth of the RTP stream due to header

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   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.  In other words a sender SHALL NOT
   increase the total bit-rate when adding redundancy in response to
   packet loss, and needs instead to adjust it down in accordance to the
   congestion control algorithm being run.  Thus when adding redundancy
   the media bit-rate will generally be needed to reduced to free up the
   bit-rate that is used for redundancy.

   The CBR signalling parameter allows a receiver to lock down a RTP
   payload type to use a single encoding rate.  As this prevents the
   codec rate from being lowered when congestion is experienced, the
   sender is constrained to either change the packetization or abort the
   transmission.  Since these responses to congestion are severely
   limited, implementations SHOULD NOT use the CBR parameter unless they
   are interacting with a device that cannot support variable bit rate
   (e.g. a gateway to H.320 systems).  When using CBR mode, a receiver
   MUST monitor the packet loss rate to ensure congestion is not caused,
   following the guidelines in Section 2 of RFC 3551.

10.  Security Considerations

   RTP packets using the payload format defined in this specification
   are subject to the general security considerations discussed in RTP
   [RFC3550] and any applicable profile such as AVP [RFC3551] or SAVP
   [RFC3711].  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 [RFC3711], MAY be used.

   This payload format and the G.719 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.
   The payload format or the codec data does not contain any type of
   active content such as scripts.

10.1.  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

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   useful to a third party, for example, for quality monitoring.
   However, as there currently don't exist any mechanism supporting
   differential protection, this behavior isn't expected to be supported
   and requirement of the audio data will be what governs the protection
   of the RTP payload.

   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.

10.2.  Authentication and Integrity

   To authenticate the sender of the audio-stream, an external mechanism
   MUST be used.  It is RECOMMENDED that such a mechanism protects both
   the complete RTP header and the payload (audio 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.

11.  Acknowledgements

   The authors would like to thank Roni Even and Anisse Taleb for their
   help with this draft.  We would also like to thank the people that
   has provided feedback; Colin Perkins, Mark Baker and Stephen Botzko.

12.  References

12.1.  Informative References

   [RFC2198]  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.

   [RFC2326]  Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
              Streaming Protocol (RTSP)", RFC 2326, April 1998.

   [RFC2974]  Handley, M., Perkins, C., and E. Whelan, "Session

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              Announcement Protocol", RFC 2974, October 2000.

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

   [RFC3839]  Castagno, R. and D. Singer, "MIME Type Registrations for
              3rd Generation Partnership Project (3GPP) Multimedia
              files", RFC 3839, July 2004.

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

   [RFC4337]  Y Lim and D. Singer, "MIME Type Registration for MPEG-4",
              RFC 4337, March 2006.

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

   [RFC5109]  Li, A., "RTP Payload Format for Generic Forward Error
              Correction", RFC 5109, December 2007.

12.2.  Normative References

              Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
              for Application Designers",
              draft-ietf-tsvwg-udp-guidelines-11 (work in progress),
              October 2008.

              ITU-T, "Specification : ITU-T G.719 extension for 20 kHz
              fullband audio", April 2008.

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

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with 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 S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              July 2003.

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

   [RFC5234]  Crocker, D. and P. Overell, "Augmented BNF for Syntax
              Specifications: ABNF", STD 68, RFC 5234, January 2008.

Authors' Addresses

   Magnus Westerlund
   Ericsson AB
   Torshamnsgatan 21-23
   SE-164 83 Stockholm

   Phone: +46 8 7190000

   Ingemar Johansson
   Ericsson AB
   Laboratoriegrand 11
   SE-971 28 Lulea

   Phone: +46 73 0783289

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