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Network Working Group H. Schulzrinne Request for Comments: 3551 Columbia University Obsoletes: 1890 S. Casner Category: Standards Track Packet Design

                                                             July 2003
            RTP Profile for Audio and Video Conferences
                        with Minimal Control

Status of this Memo

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

Copyright Notice

 Copyright (C) The Internet Society (2003).  All Rights Reserved.

Abstract

 This document describes a profile called "RTP/AVP" for the use of the
 real-time transport protocol (RTP), version 2, and the associated
 control protocol, RTCP, within audio and video multiparticipant
 conferences with minimal control.  It provides interpretations of
 generic fields within the RTP specification suitable for audio and
 video conferences.  In particular, this document defines a set of
 default mappings from payload type numbers to encodings.
 This document also describes how audio and video data may be carried
 within RTP.  It defines a set of standard encodings and their names
 when used within RTP.  The descriptions provide pointers to reference
 implementations and the detailed standards.  This document is meant
 as an aid for implementors of audio, video and other real-time
 multimedia applications.
 This memorandum obsoletes RFC 1890.  It is mostly backwards-
 compatible except for functions removed because two interoperable
 implementations were not found.  The additions to RFC 1890 codify
 existing practice in the use of payload formats under this profile
 and include new payload formats defined since RFC 1890 was published.

Schulzrinne & Casner Standards Track [Page 1] RFC 3551 RTP A/V Profile July 2003

Table of Contents

 1.  Introduction .................................................  3
     1.1  Terminology .............................................  3
 2.  RTP and RTCP Packet Forms and Protocol Behavior ..............  4
 3.  Registering Additional Encodings .............................  6
 4.  Audio ........................................................  8
     4.1  Encoding-Independent Rules ..............................  8
     4.2  Operating Recommendations ...............................  9
     4.3  Guidelines for Sample-Based Audio Encodings ............. 10
     4.4  Guidelines for Frame-Based Audio Encodings .............. 11
     4.5  Audio Encodings ......................................... 12
          4.5.1   DVI4 ............................................ 13
          4.5.2   G722 ............................................ 14
          4.5.3   G723 ............................................ 14
          4.5.4   G726-40, G726-32, G726-24, and G726-16 .......... 18
          4.5.5   G728 ............................................ 19
          4.5.6   G729 ............................................ 20
          4.5.7   G729D and G729E ................................. 22
          4.5.8   GSM ............................................. 24
          4.5.9   GSM-EFR ......................................... 27
          4.5.10  L8 .............................................. 27
          4.5.11  L16 ............................................. 27
          4.5.12  LPC ............................................. 27
          4.5.13  MPA ............................................. 28
          4.5.14  PCMA and PCMU ................................... 28
          4.5.15  QCELP ........................................... 28
          4.5.16  RED ............................................. 29
          4.5.17  VDVI ............................................ 29
 5.  Video ........................................................ 30
     5.1  CelB .................................................... 30
     5.2  JPEG .................................................... 30
     5.3  H261 .................................................... 30
     5.4  H263 .................................................... 31
     5.5  H263-1998 ............................................... 31
     5.6  MPV ..................................................... 31
     5.7  MP2T .................................................... 31
     5.8  nv ...................................................... 32
 6.  Payload Type Definitions ..................................... 32
 7.  RTP over TCP and Similar Byte Stream Protocols ............... 34
 8.  Port Assignment .............................................. 34
 9.  Changes from RFC 1890 ........................................ 35
 10. Security Considerations ...................................... 38
 11. IANA Considerations .......................................... 39
 12. References ................................................... 39
     12.1 Normative References .................................... 39
     12.2 Informative References .................................. 39
 13. Current Locations of Related Resources ....................... 41

Schulzrinne & Casner Standards Track [Page 2] RFC 3551 RTP A/V Profile July 2003

 14. Acknowledgments .............................................. 42
 15. Intellectual Property Rights Statement ....................... 43
 16. Authors' Addresses ........................................... 43
 17. Full Copyright Statement ..................................... 44

1. Introduction

 This profile defines aspects of RTP left unspecified in the RTP
 Version 2 protocol definition (RFC 3550) [1].  This profile is
 intended for the use within audio and video conferences with minimal
 session control.  In particular, no support for the negotiation of
 parameters or membership control is provided.  The profile is
 expected to be useful in sessions where no negotiation or membership
 control are used (e.g., using the static payload types and the
 membership indications provided by RTCP), but this profile may also
 be useful in conjunction with a higher-level control protocol.
 Use of this profile may be implicit in the use of the appropriate
 applications; there may be no explicit indication by port number,
 protocol identifier or the like.  Applications such as session
 directories may use the name for this profile specified in Section
 11.
 Other profiles may make different choices for the items specified
 here.
 This document also defines a set of encodings and payload formats for
 audio and video.  These payload format descriptions are included here
 only as a matter of convenience since they are too small to warrant
 separate documents.  Use of these payload formats is NOT REQUIRED to
 use this profile.  Only the binding of some of the payload formats to
 static payload type numbers in Tables 4 and 5 is normative.

1.1 Terminology

 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
 document are to be interpreted as described in RFC 2119 [2] and
 indicate requirement levels for implementations compliant with this
 RTP profile.
 This document defines the term media type as dividing encodings of
 audio and video content into three classes: audio, video and
 audio/video (interleaved).

Schulzrinne & Casner Standards Track [Page 3] RFC 3551 RTP A/V Profile July 2003

2. RTP and RTCP Packet Forms and Protocol Behavior

 The section "RTP Profiles and Payload Format Specifications" of RFC
 3550 enumerates a number of items that can be specified or modified
 in a profile.  This section addresses these items.  Generally, this
 profile follows the default and/or recommended aspects of the RTP
 specification.
 RTP data header: The standard format of the fixed RTP data
    header is used (one marker bit).
 Payload types: Static payload types are defined in Section 6.
 RTP data header additions: No additional fixed fields are
    appended to the RTP data header.
 RTP data header extensions: No RTP header extensions are
    defined, but applications operating under this profile MAY use
    such extensions.  Thus, applications SHOULD NOT assume that the
    RTP header X bit is always zero and SHOULD be prepared to ignore
    the header extension.  If a header extension is defined in the
    future, that definition MUST specify the contents of the first 16
    bits in such a way that multiple different extensions can be
    identified.
 RTCP packet types: No additional RTCP packet types are defined
    by this profile specification.
 RTCP report interval: The suggested constants are to be used for
    the RTCP report interval calculation.  Sessions operating under
    this profile MAY specify a separate parameter for the RTCP traffic
    bandwidth rather than using the default fraction of the session
    bandwidth.  The RTCP traffic bandwidth MAY be divided into two
    separate session parameters for those participants which are
    active data senders and those which are not.  Following the
    recommendation in the RTP specification [1] that 1/4 of the RTCP
    bandwidth be dedicated to data senders, the RECOMMENDED default
    values for these two parameters would be 1.25% and 3.75%,
    respectively.  For a particular session, the RTCP bandwidth for
    non-data-senders MAY be set to zero when operating on
    unidirectional links or for sessions that don't require feedback
    on the quality of reception.  The RTCP bandwidth for data senders
    SHOULD be kept non-zero so that sender reports can still be sent
    for inter-media synchronization and to identify the source by
    CNAME.  The means by which the one or two session parameters for
    RTCP bandwidth are specified is beyond the scope of this memo.

Schulzrinne & Casner Standards Track [Page 4] RFC 3551 RTP A/V Profile July 2003

 SR/RR extension: No extension section is defined for the RTCP SR
    or RR packet.
 SDES use: Applications MAY use any of the SDES items described
    in the RTP specification.  While CNAME information MUST be sent
    every reporting interval, other items SHOULD only be sent every
    third reporting interval, with NAME sent seven out of eight times
    within that slot and the remaining SDES items cyclically taking up
    the eighth slot, as defined in Section 6.2.2 of the RTP
    specification.  In other words, NAME is sent in RTCP packets 1, 4,
    7, 10, 13, 16, 19, while, say, EMAIL is used in RTCP packet 22.
 Security: The RTP default security services are also the default
    under this profile.
 String-to-key mapping: No mapping is specified by this profile.
 Congestion: RTP and this profile may be used in the context of
    enhanced network service, for example, through Integrated Services
    (RFC 1633) [4] or Differentiated Services (RFC 2475) [5], or they
    may be used with best effort service.
    If enhanced service is being used, RTP receivers SHOULD monitor
    packet loss to ensure that the service that was requested is
    actually being delivered.  If it is not, then they SHOULD assume
    that they are receiving best-effort service and behave
    accordingly.
    If best-effort service is being used, RTP receivers SHOULD monitor
    packet loss to ensure that the packet loss rate is within
    acceptable parameters.  Packet loss is considered acceptable if a
    TCP flow across the same network path and experiencing the same
    network conditions would achieve an average throughput, measured
    on a reasonable timescale, that is not less than the RTP flow is
    achieving.  This condition can be satisfied by implementing
    congestion control mechanisms to adapt the transmission rate (or
    the number of layers subscribed for a layered multicast session),
    or by arranging for a receiver to leave the session if the loss
    rate is unacceptably high.
    The comparison to TCP cannot be specified exactly, but is intended
    as an "order-of-magnitude" comparison in timescale and throughput.
    The timescale on which TCP throughput is measured is the round-
    trip time of the connection.  In essence, this requirement states
    that it is not acceptable to deploy an application (using RTP or
    any other transport protocol) on the best-effort Internet which
    consumes bandwidth arbitrarily and does not compete fairly with
    TCP within an order of magnitude.

Schulzrinne & Casner Standards Track [Page 5] RFC 3551 RTP A/V Profile July 2003

 Underlying protocol: The profile specifies the use of RTP over
    unicast and multicast UDP as well as TCP.  (This does not preclude
    the use of these definitions when RTP is carried by other lower-
    layer protocols.)
 Transport mapping: The standard mapping of RTP and RTCP to
    transport-level addresses is used.
 Encapsulation: This profile leaves to applications the
    specification of RTP encapsulation in protocols other than UDP.

3. Registering Additional Encodings

 This profile lists a set of encodings, each of which is comprised of
 a particular media data compression or representation plus a payload
 format for encapsulation within RTP.  Some of those payload formats
 are specified here, while others are specified in separate RFCs.  It
 is expected that additional encodings beyond the set listed here will
 be created in the future and specified in additional payload format
 RFCs.
 This profile also assigns to each encoding a short name which MAY be
 used by higher-level control protocols, such as the Session
 Description Protocol (SDP), RFC 2327 [6], to identify encodings
 selected for a particular RTP session.
 In some contexts it may be useful to refer to these encodings in the
 form of a MIME content-type.  To facilitate this, RFC 3555 [7]
 provides registrations for all of the encodings names listed here as
 MIME subtype names under the "audio" and "video" MIME types through
 the MIME registration procedure as specified in RFC 2048 [8].
 Any additional encodings specified for use under this profile (or
 others) may also be assigned names registered as MIME subtypes with
 the Internet Assigned Numbers Authority (IANA).  This registry
 provides a means to insure that the names assigned to the additional
 encodings are kept unique.  RFC 3555 specifies the information that
 is required for the registration of RTP encodings.
 In addition to assigning names to encodings, this profile also
 assigns static RTP payload type numbers to some of them.  However,
 the payload type number space is relatively small and cannot
 accommodate assignments for all existing and future encodings.
 During the early stages of RTP development, it was necessary to use
 statically assigned payload types because no other mechanism had been
 specified to bind encodings to payload types.  It was anticipated
 that non-RTP means beyond the scope of this memo (such as directory
 services or invitation protocols) would be specified to establish a

Schulzrinne & Casner Standards Track [Page 6] RFC 3551 RTP A/V Profile July 2003

 dynamic mapping between a payload type and an encoding.  Now,
 mechanisms for defining dynamic payload type bindings have been
 specified in the Session Description Protocol (SDP) and in other
 protocols such as ITU-T Recommendation H.323/H.245.  These mechanisms
 associate the registered name of the encoding/payload format, along
 with any additional required parameters, such as the RTP timestamp
 clock rate and number of channels, with a payload type number.  This
 association is effective only for the duration of the RTP session in
 which the dynamic payload type binding is made.  This association
 applies only to the RTP session for which it is made, thus the
 numbers can be re-used for different encodings in different sessions
 so the number space limitation is avoided.
 This profile reserves payload type numbers in the range 96-127
 exclusively for dynamic assignment.  Applications SHOULD first use
 values in this range for dynamic payload types.  Those applications
 which need to define more than 32 dynamic payload types MAY bind
 codes below 96, in which case it is RECOMMENDED that unassigned
 payload type numbers be used first.  However, the statically assigned
 payload types are default bindings and MAY be dynamically bound to
 new encodings if needed.  Redefining payload types below 96 may cause
 incorrect operation if an attempt is made to join a session without
 obtaining session description information that defines the dynamic
 payload types.
 Dynamic payload types SHOULD NOT be used without a well-defined
 mechanism to indicate the mapping.  Systems that expect to
 interoperate with others operating under this profile SHOULD NOT make
 their own assignments of proprietary encodings to particular, fixed
 payload types.
 This specification establishes the policy that no additional static
 payload types will be assigned beyond the ones defined in this
 document.  Establishing this policy avoids the problem of trying to
 create a set of criteria for accepting static assignments and
 encourages the implementation and deployment of the dynamic payload
 type mechanisms.
 The final set of static payload type assignments is provided in
 Tables 4 and 5.

Schulzrinne & Casner Standards Track [Page 7] RFC 3551 RTP A/V Profile July 2003

4. Audio

4.1 Encoding-Independent Rules

 Since the ability to suppress silence is one of the primary
 motivations for using packets to transmit voice, the RTP header
 carries both a sequence number and a timestamp to allow a receiver to
 distinguish between lost packets and periods of time when no data was
 transmitted.  Discontiguous transmission (silence suppression) MAY be
 used with any audio payload format.  Receivers MUST assume that
 senders may suppress silence unless this is restricted by signaling
 specified elsewhere.  (Even if the transmitter does not suppress
 silence, the receiver should be prepared to handle periods when no
 data is present since packets may be lost.)
 Some payload formats (see Sections 4.5.3 and 4.5.6) define a "silence
 insertion descriptor" or "comfort noise" frame to specify parameters
 for artificial noise that may be generated during a period of silence
 to approximate the background noise at the source.  For other payload
 formats, a generic Comfort Noise (CN) payload format is specified in
 RFC 3389 [9].  When the CN payload format is used with another
 payload format, different values in the RTP payload type field
 distinguish comfort-noise packets from those of the selected payload
 format.
 For applications which send either no packets or occasional comfort-
 noise packets during silence, the first packet of a talkspurt, that
 is, the first packet after a silence period during which packets have
 not been transmitted contiguously, SHOULD be distinguished by setting
 the marker bit in the RTP data header to one.  The marker bit in all
 other packets is zero.  The beginning of a talkspurt MAY be used to
 adjust the playout delay to reflect changing network delays.
 Applications without silence suppression MUST set the marker bit to
 zero.
 The RTP clock rate used for generating the RTP timestamp is
 independent of the number of channels and the encoding; it usually
 equals the number of sampling periods per second.  For N-channel
 encodings, each sampling period (say, 1/8,000 of a second) generates
 N samples.  (This terminology is standard, but somewhat confusing, as
 the total number of samples generated per second is then the sampling
 rate times the channel count.)
 If multiple audio channels are used, channels are numbered left-to-
 right, starting at one.  In RTP audio packets, information from
 lower-numbered channels precedes that from higher-numbered channels.

Schulzrinne & Casner Standards Track [Page 8] RFC 3551 RTP A/V Profile July 2003

 For more than two channels, the convention followed by the AIFF-C
 audio interchange format SHOULD be followed [3], using the following
 notation, unless some other convention is specified for a particular
 encoding or payload format:
    l  left
    r  right
    c  center
    S  surround
    F  front
    R  rear
    channels  description  channel
                              1     2   3   4   5   6
    _________________________________________________
    2         stereo          l     r
    3                         l     r   c
    4                         l     c   r   S
    5                        Fl     Fr  Fc  Sl  Sr
    6                         l     lc  c   r   rc  S
       Note: RFC 1890 defined two conventions for the ordering of four
       audio channels.  Since the ordering is indicated implicitly by
       the number of channels, this was ambiguous.  In this revision,
       the order described as "quadrophonic" has been eliminated to
       remove the ambiguity.  This choice was based on the observation
       that quadrophonic consumer audio format did not become popular
       whereas surround-sound subsequently has.
 Samples for all channels belonging to a single sampling instant MUST
 be within the same packet.  The interleaving of samples from
 different channels depends on the encoding.  General guidelines are
 given in Section 4.3 and 4.4.
 The sampling frequency SHOULD be drawn from the set:  8,000, 11,025,
 16,000, 22,050, 24,000, 32,000, 44,100 and 48,000 Hz.  (Older Apple
 Macintosh computers had a native sample rate of 22,254.54 Hz, which
 can be converted to 22,050 with acceptable quality by dropping 4
 samples in a 20 ms frame.)  However, most audio encodings are defined
 for a more restricted set of sampling frequencies.  Receivers SHOULD
 be prepared to accept multi-channel audio, but MAY choose to only
 play a single channel.

4.2 Operating Recommendations

 The following recommendations are default operating parameters.
 Applications SHOULD be prepared to handle other values.  The ranges
 given are meant to give guidance to application writers, allowing a

Schulzrinne & Casner Standards Track [Page 9] RFC 3551 RTP A/V Profile July 2003

 set of applications conforming to these guidelines to interoperate
 without additional negotiation.  These guidelines are not intended to
 restrict operating parameters for applications that can negotiate a
 set of interoperable parameters, e.g., through a conference control
 protocol.
 For packetized audio, the default packetization interval SHOULD have
 a duration of 20 ms or one frame, whichever is longer, unless
 otherwise noted in Table 1 (column "ms/packet").  The packetization
 interval determines the minimum end-to-end delay; longer packets
 introduce less header overhead but higher delay and make packet loss
 more noticeable.  For non-interactive applications such as lectures
 or for links with severe bandwidth constraints, a higher
 packetization delay MAY be used.  A receiver SHOULD accept packets
 representing between 0 and 200 ms of audio data.  (For framed audio
 encodings, a receiver SHOULD accept packets with a number of frames
 equal to 200 ms divided by the frame duration, rounded up.)  This
 restriction allows reasonable buffer sizing for the receiver.

4.3 Guidelines for Sample-Based Audio Encodings

 In sample-based encodings, each audio sample is represented by a
 fixed number of bits.  Within the compressed audio data, codes for
 individual samples may span octet boundaries.  An RTP audio packet
 may contain any number of audio samples, subject to the constraint
 that the number of bits per sample times the number of samples per
 packet yields an integral octet count.  Fractional encodings produce
 less than one octet per sample.
 The duration of an audio packet is determined by the number of
 samples in the packet.
 For sample-based encodings producing one or more octets per sample,
 samples from different channels sampled at the same sampling instant
 SHOULD be packed in consecutive octets.  For example, for a two-
 channel encoding, the octet sequence is (left channel, first sample),
 (right channel, first sample), (left channel, second sample), (right
 channel, second sample), ....  For multi-octet encodings, octets
 SHOULD be transmitted in network byte order (i.e., most significant
 octet first).
 The packing of sample-based encodings producing less than one octet
 per sample is encoding-specific.
 The RTP timestamp reflects the instant at which the first sample in
 the packet was sampled, that is, the oldest information in the
 packet.

Schulzrinne & Casner Standards Track [Page 10] RFC 3551 RTP A/V Profile July 2003

4.4 Guidelines for Frame-Based Audio Encodings

 Frame-based encodings encode a fixed-length block of audio into
 another block of compressed data, typically also of fixed length.
 For frame-based encodings, the sender MAY choose to combine several
 such frames into a single RTP packet.  The receiver can tell the
 number of frames contained in an RTP packet, if all the frames have
 the same length, by dividing the RTP payload length by the audio
 frame size which is defined as part of the encoding.  This does not
 work when carrying frames of different sizes unless the frame sizes
 are relatively prime.  If not, the frames MUST indicate their size.
 For frame-based codecs, the channel order is defined for the whole
 block.  That is, for two-channel audio, right and left samples SHOULD
 be coded independently, with the encoded frame for the left channel
 preceding that for the right channel.
 All frame-oriented audio codecs SHOULD be able to encode and decode
 several consecutive frames within a single packet.  Since the frame
 size for the frame-oriented codecs is given, there is no need to use
 a separate designation for the same encoding, but with different
 number of frames per packet.
 RTP packets SHALL contain a whole number of frames, with frames
 inserted according to age within a packet, so that the oldest frame
 (to be played first) occurs immediately after the RTP packet header.
 The RTP timestamp reflects the instant at which the first sample in
 the first frame was sampled, that is, the oldest information in the
 packet.

Schulzrinne & Casner Standards Track [Page 11] RFC 3551 RTP A/V Profile July 2003

4.5 Audio Encodings

 name of                              sampling              default
 encoding  sample/frame  bits/sample      rate  ms/frame  ms/packet
 __________________________________________________________________
 DVI4      sample        4                var.                   20
 G722      sample        8              16,000                   20
 G723      frame         N/A             8,000        30         30
 G726-40   sample        5               8,000                   20
 G726-32   sample        4               8,000                   20
 G726-24   sample        3               8,000                   20
 G726-16   sample        2               8,000                   20
 G728      frame         N/A             8,000       2.5         20
 G729      frame         N/A             8,000        10         20
 G729D     frame         N/A             8,000        10         20
 G729E     frame         N/A             8,000        10         20
 GSM       frame         N/A             8,000        20         20
 GSM-EFR   frame         N/A             8,000        20         20
 L8        sample        8                var.                   20
 L16       sample        16               var.                   20
 LPC       frame         N/A             8,000        20         20
 MPA       frame         N/A              var.      var.
 PCMA      sample        8                var.                   20
 PCMU      sample        8                var.                   20
 QCELP     frame         N/A             8,000        20         20
 VDVI      sample        var.             var.                   20
 Table 1: Properties of Audio Encodings (N/A: not applicable; var.:
          variable)
 The characteristics of the audio encodings described in this document
 are shown in Table 1; they are listed in order of their payload type
 in Table 4.  While most audio codecs are only specified for a fixed
 sampling rate, some sample-based algorithms (indicated by an entry of
 "var." in the sampling rate column of Table 1) may be used with
 different sampling rates, resulting in different coded bit rates.
 When used with a sampling rate other than that for which a static
 payload type is defined, non-RTP means beyond the scope of this memo
 MUST be used to define a dynamic payload type and MUST indicate the
 selected RTP timestamp clock rate, which is usually the same as the
 sampling rate for audio.

Schulzrinne & Casner Standards Track [Page 12] RFC 3551 RTP A/V Profile July 2003

4.5.1 DVI4

 DVI4 uses an adaptive delta pulse code modulation (ADPCM) encoding
 scheme that was specified by the Interactive Multimedia Association
 (IMA) as the "IMA ADPCM wave type".  However, the encoding defined
 here as DVI4 differs in three respects from the IMA specification:
 o  The RTP DVI4 header contains the predicted value rather than the
    first sample value contained the IMA ADPCM block header.
 o  IMA ADPCM blocks contain an odd number of samples, since the first
    sample of a block is contained just in the header (uncompressed),
    followed by an even number of compressed samples.  DVI4 has an
    even number of compressed samples only, using the `predict' word
    from the header to decode the first sample.
 o  For DVI4, the 4-bit samples are packed with the first sample in
    the four most significant bits and the second sample in the four
    least significant bits.  In the IMA ADPCM codec, the samples are
    packed in the opposite order.
 Each packet contains a single DVI block.  This profile only defines
 the 4-bit-per-sample version, while IMA also specified a 3-bit-per-
 sample encoding.
 The "header" word for each channel has the following structure:
    int16  predict;  /* predicted value of first sample
                        from the previous block (L16 format) */
    u_int8 index;    /* current index into stepsize table */
    u_int8 reserved; /* set to zero by sender, ignored by receiver */
 Each octet following the header contains two 4-bit samples, thus the
 number of samples per packet MUST be even because there is no means
 to indicate a partially filled last octet.
 Packing of samples for multiple channels is for further study.
 The IMA ADPCM algorithm was described in the document IMA Recommended
 Practices for Enhancing Digital Audio Compatibility in Multimedia
 Systems (version 3.0).  However, the Interactive Multimedia
 Association ceased operations in 1997.  Resources for an archived
 copy of that document and a software implementation of the RTP DVI4
 encoding are listed in Section 13.

Schulzrinne & Casner Standards Track [Page 13] RFC 3551 RTP A/V Profile July 2003

4.5.2 G722

 G722 is specified in ITU-T Recommendation G.722, "7 kHz audio-coding
 within 64 kbit/s".  The G.722 encoder produces a stream of octets,
 each of which SHALL be octet-aligned in an RTP packet.  The first bit
 transmitted in the G.722 octet, which is the most significant bit of
 the higher sub-band sample, SHALL correspond to the most significant
 bit of the octet in the RTP packet.
 Even though the actual sampling rate for G.722 audio is 16,000 Hz,
 the RTP clock rate for the G722 payload format is 8,000 Hz because
 that value was erroneously assigned in RFC 1890 and must remain
 unchanged for backward compatibility.  The octet rate or sample-pair
 rate is 8,000 Hz.

4.5.3 G723

 G723 is specified in ITU Recommendation G.723.1, "Dual-rate speech
 coder for multimedia communications transmitting at 5.3 and 6.3
 kbit/s".  The G.723.1 5.3/6.3 kbit/s codec was defined by the ITU-T
 as a mandatory codec for ITU-T H.324 GSTN videophone terminal
 applications.  The algorithm has a floating point specification in
 Annex B to G.723.1, a silence compression algorithm in Annex A to
 G.723.1 and a scalable channel coding scheme for wireless
 applications in G.723.1 Annex C.
 This Recommendation specifies a coded representation that can be used
 for compressing the speech signal component of multi-media services
 at a very low bit rate.  Audio is encoded in 30 ms frames, with an
 additional delay of 7.5 ms due to look-ahead.  A G.723.1 frame can be
 one of three sizes:  24 octets (6.3 kb/s frame), 20 octets (5.3 kb/s
 frame), or 4 octets.  These 4-octet frames are called SID frames
 (Silence Insertion Descriptor) and are used to specify comfort noise
 parameters.  There is no restriction on how 4, 20, and 24 octet
 frames are intermixed.  The least significant two bits of the first
 octet in the frame determine the frame size and codec type:
       bits  content                      octets/frame
       00    high-rate speech (6.3 kb/s)            24
       01    low-rate speech  (5.3 kb/s)            20
       10    SID frame                               4
       11    reserved

Schulzrinne & Casner Standards Track [Page 14] RFC 3551 RTP A/V Profile July 2003

 It is possible to switch between the two rates at any 30 ms frame
 boundary.  Both (5.3 kb/s and 6.3 kb/s) rates are a mandatory part of
 the encoder and decoder.  Receivers MUST accept both data rates and
 MUST accept SID frames unless restriction of these capabilities has
 been signaled.  The MIME registration for G723 in RFC 3555 [7]
 specifies parameters that MAY be used with MIME or SDP to restrict to
 a single data rate or to restrict the use of SID frames.  This coder
 was optimized to represent speech with near-toll quality at the above
 rates using a limited amount of complexity.
 The packing of the encoded bit stream into octets and the
 transmission order of the octets is specified in Rec. G.723.1 and is
 the same as that produced by the G.723 C code reference
 implementation.  For the 6.3 kb/s data rate, this packing is
 illustrated as follows, where the header (HDR) bits are always "0 0"
 as shown in Fig. 1 to indicate operation at 6.3 kb/s, and the Z bit
 is always set to zero.  The diagrams show the bit packing in "network
 byte order", also known as big-endian order.  The bits of each 32-bit
 word are numbered 0 to 31, with the most significant bit on the left
 and numbered 0.  The octets (bytes) of each word are transmitted most
 significant octet first.  The bits of each data field are numbered in
 the order of the bit stream representation of the encoding (least
 significant bit first).  The vertical bars indicate the boundaries
 between field fragments.

Schulzrinne & Casner Standards Track [Page 15] RFC 3551 RTP A/V Profile July 2003

  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
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |    LPC    |HDR|      LPC      |      LPC      |    ACL0   |LPC|
 |           |   |               |               |           |   |
 |0 0 0 0 0 0|0 0|1 1 1 1 0 0 0 0|2 2 1 1 1 1 1 1|0 0 0 0 0 0|2 2|
 |5 4 3 2 1 0|   |3 2 1 0 9 8 7 6|1 0 9 8 7 6 5 4|5 4 3 2 1 0|3 2|
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |  ACL2   |ACL|A| GAIN0 |ACL|ACL|    GAIN0      |    GAIN1      |
 |         | 1 |C|       | 3 | 2 |               |               |
 |0 0 0 0 0|0 0|0|0 0 0 0|0 0|0 0|1 1 0 0 0 0 0 0|0 0 0 0 0 0 0 0|
 |4 3 2 1 0|1 0|6|3 2 1 0|1 0|6 5|1 0 9 8 7 6 5 4|7 6 5 4 3 2 1 0|
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | GAIN2 | GAIN1 |     GAIN2     |     GAIN3     | GRID  | GAIN3 |
 |       |       |               |               |       |       |
 |0 0 0 0|1 1 0 0|1 1 0 0 0 0 0 0|0 0 0 0 0 0 0 0|0 0 0 0|1 1 0 0|
 |3 2 1 0|1 0 9 8|1 0 9 8 7 6 5 4|7 6 5 4 3 2 1 0|3 2 1 0|1 0 9 8|
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |   MSBPOS    |Z|POS|  MSBPOS   |     POS0      |POS|   POS0    |
 |             | | 0 |           |               | 1 |           |
 |0 0 0 0 0 0 0|0|0 0|1 1 1 0 0 0|0 0 0 0 0 0 0 0|0 0|1 1 1 1 1 1|
 |6 5 4 3 2 1 0| |1 0|2 1 0 9 8 7|9 8 7 6 5 4 3 2|1 0|5 4 3 2 1 0|
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |     POS1      | POS2  | POS1  |     POS2      | POS3  | POS2  |
 |               |       |       |               |       |       |
 |0 0 0 0 0 0 0 0|0 0 0 0|1 1 1 1|1 1 0 0 0 0 0 0|0 0 0 0|1 1 1 1|
 |9 8 7 6 5 4 3 2|3 2 1 0|3 2 1 0|1 0 9 8 7 6 5 4|3 2 1 0|5 4 3 2|
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |     POS3      |   PSIG0   |POS|PSIG2|  PSIG1  |  PSIG3  |PSIG2|
 |               |           | 3 |     |         |         |     |
 |1 1 0 0 0 0 0 0|0 0 0 0 0 0|1 1|0 0 0|0 0 0 0 0|0 0 0 0 0|0 0 0|
 |1 0 9 8 7 6 5 4|5 4 3 2 1 0|3 2|2 1 0|4 3 2 1 0|4 3 2 1 0|5 4 3|
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
                Figure 1: G.723 (6.3 kb/s) bit packing
 For the 5.3 kb/s data rate, the header (HDR) bits are always "0 1",
 as shown in Fig. 2, to indicate operation at 5.3 kb/s.

Schulzrinne & Casner Standards Track [Page 16] RFC 3551 RTP A/V Profile July 2003

  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
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |    LPC    |HDR|      LPC      |      LPC      |   ACL0    |LPC|
 |           |   |               |               |           |   |
 |0 0 0 0 0 0|0 1|1 1 1 1 0 0 0 0|2 2 1 1 1 1 1 1|0 0 0 0 0 0|2 2|
 |5 4 3 2 1 0|   |3 2 1 0 9 8 7 6|1 0 9 8 7 6 5 4|5 4 3 2 1 0|3 2|
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |  ACL2   |ACL|A| GAIN0 |ACL|ACL|     GAIN0     |     GAIN1     |
 |         | 1 |C|       | 3 | 2 |               |               |
 |0 0 0 0 0|0 0|0|0 0 0 0|0 0|0 0|1 1 0 0 0 0 0 0|0 0 0 0 0 0 0 0|
 |4 3 2 1 0|1 0|6|3 2 1 0|1 0|6 5|1 0 9 8 7 6 5 4|7 6 5 4 3 2 1 0|
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | GAIN2 | GAIN1 |     GAIN2     |    GAIN3      | GRID  | GAIN3 |
 |       |       |               |               |       |       |
 |0 0 0 0|1 1 0 0|1 1 0 0 0 0 0 0|0 0 0 0 0 0 0 0|0 0 0 0|1 1 0 0|
 |3 2 1 0|1 0 9 8|1 0 9 8 7 6 5 4|7 6 5 4 3 2 1 0|4 3 2 1|1 0 9 8|
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |     POS0      | POS1  | POS0  |     POS1      |     POS2      |
 |               |       |       |               |               |
 |0 0 0 0 0 0 0 0|0 0 0 0|1 1 0 0|1 1 0 0 0 0 0 0|0 0 0 0 0 0 0 0|
 |7 6 5 4 3 2 1 0|3 2 1 0|1 0 9 8|1 0 9 8 7 6 5 4|7 6 5 4 3 2 1 0|
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | POS3  | POS2  |     POS3      | PSIG1 | PSIG0 | PSIG3 | PSIG2 |
 |       |       |               |       |       |       |       |
 |0 0 0 0|1 1 0 0|1 1 0 0 0 0 0 0|0 0 0 0|0 0 0 0|0 0 0 0|0 0 0 0|
 |3 2 1 0|1 0 9 8|1 0 9 8 7 6 5 4|3 2 1 0|3 2 1 0|3 2 1 0|3 2 1 0|
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
                Figure 2: G.723 (5.3 kb/s) bit packing
 The packing of G.723.1 SID (silence) frames, which are indicated by
 the header (HDR) bits having the pattern "1 0", is depicted in Fig.
 3.
  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
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |    LPC    |HDR|      LPC      |      LPC      |   GAIN    |LPC|
 |           |   |               |               |           |   |
 |0 0 0 0 0 0|1 0|1 1 1 1 0 0 0 0|2 2 1 1 1 1 1 1|0 0 0 0 0 0|2 2|
 |5 4 3 2 1 0|   |3 2 1 0 9 8 7 6|1 0 9 8 7 6 5 4|5 4 3 2 1 0|3 2|
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
                 Figure 3: G.723 SID mode bit packing

Schulzrinne & Casner Standards Track [Page 17] RFC 3551 RTP A/V Profile July 2003

4.5.4 G726-40, G726-32, G726-24, and G726-16

 ITU-T Recommendation G.726 describes, among others, the algorithm
 recommended for conversion of a single 64 kbit/s A-law or mu-law PCM
 channel encoded at 8,000 samples/sec to and from a 40, 32, 24, or 16
 kbit/s channel.  The conversion is applied to the PCM stream using an
 Adaptive Differential Pulse Code Modulation (ADPCM) transcoding
 technique.  The ADPCM representation consists of a series of
 codewords with a one-to-one correspondence to the samples in the PCM
 stream.  The G726 data rates of 40, 32, 24, and 16 kbit/s have
 codewords of 5, 4, 3, and 2 bits, respectively.
 The 16 and 24 kbit/s encodings do not provide toll quality speech.
 They are designed for used in overloaded Digital Circuit
 Multiplication Equipment (DCME).  ITU-T G.726 recommends that the 16
 and 24 kbit/s encodings should be alternated with higher data rate
 encodings to provide an average sample size of between 3.5 and 3.7
 bits per sample.
 The encodings of G.726 are here denoted as G726-40, G726-32, G726-24,
 and G726-16.  Prior to 1990, G721 described the 32 kbit/s ADPCM
 encoding, and G723 described the 40, 32, and 16 kbit/s encodings.
 Thus, G726-32 designates the same algorithm as G721 in RFC 1890.
 A stream of G726 codewords contains no information on the encoding
 being used, therefore transitions between G726 encoding types are not
 permitted within a sequence of packed codewords.  Applications MUST
 determine the encoding type of packed codewords from the RTP payload
 identifier.
 No payload-specific header information SHALL be included as part of
 the audio data.  A stream of G726 codewords MUST be packed into
 octets as follows:  the first codeword is placed into the first octet
 such that the least significant bit of the codeword aligns with the
 least significant bit in the octet, the second codeword is then
 packed so that its least significant bit coincides with the least
 significant unoccupied bit in the octet.  When a complete codeword
 cannot be placed into an octet, the bits overlapping the octet
 boundary are placed into the least significant bits of the next
 octet.  Packing MUST end with a completely packed final octet.  The
 number of codewords packed will therefore be a multiple of 8, 2, 8,
 and 4 for G726-40, G726-32, G726-24, and G726-16, respectively.  An
 example of the packing scheme for G726-32 codewords is as shown,
 where bit 7 is the least significant bit of the first octet, and bit
 A3 is the least significant bit of the first codeword:

Schulzrinne & Casner Standards Track [Page 18] RFC 3551 RTP A/V Profile July 2003

        0                   1
        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
       |B B B B|A A A A|D D D D|C C C C| ...
       |0 1 2 3|0 1 2 3|0 1 2 3|0 1 2 3|
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
 An example of the packing scheme for G726-24 codewords follows, where
 again bit 7 is the least significant bit of the first octet, and bit
 A2 is the least significant bit of the first codeword:
        0                   1                   2
        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
       |C C|B B B|A A A|F|E E E|D D D|C|H H H|G G G|F F| ...
       |1 2|0 1 2|0 1 2|2|0 1 2|0 1 2|0|0 1 2|0 1 2|0 1|
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
 Note that the "little-endian" direction in which samples are packed
 into octets in the G726-16, -24, -32 and -40 payload formats
 specified here is consistent with ITU-T Recommendation X.420, but is
 the opposite of what is specified in ITU-T Recommendation I.366.2
 Annex E for ATM AAL2 transport.  A second set of RTP payload formats
 matching the packetization of I.366.2 Annex E and identified by MIME
 subtypes AAL2-G726-16, -24, -32 and -40 will be specified in a
 separate document.

4.5.5 G728

 G728 is specified in ITU-T Recommendation G.728, "Coding of speech at
 16 kbit/s using low-delay code excited linear prediction".
 A G.278 encoder translates 5 consecutive audio samples into a 10-bit
 codebook index, resulting in a bit rate of 16 kb/s for audio sampled
 at 8,000 samples per second.  The group of five consecutive samples
 is called a vector.  Four consecutive vectors, labeled V1 to V4
 (where V1 is to be played first by the receiver), build one G.728
 frame.  The four vectors of 40 bits are packed into 5 octets, labeled
 B1 through B5.  B1 SHALL be placed first in the RTP packet.
 Referring to the figure below, the principle for bit order is
 "maintenance of bit significance".  Bits from an older vector are
 more significant than bits from newer vectors.  The MSB of the frame
 goes to the MSB of B1 and the LSB of the frame goes to LSB of B5.

Schulzrinne & Casner Standards Track [Page 19] RFC 3551 RTP A/V Profile July 2003

                 1         2         3        3
       0         0         0         0        9
       ++++++++++++++++++++++++++++++++++++++++
       <---V1---><---V2---><---V3---><---V4---> vectors
       <--B1--><--B2--><--B3--><--B4--><--B5--> octets
       <------------- frame 1 ---------------->
 In particular, B1 contains the eight most significant bits of V1,
 with the MSB of V1 being the MSB of B1.  B2 contains the two least
 significant bits of V1, the more significant of the two in its MSB,
 and the six most significant bits of V2.  B1 SHALL be placed first in
 the RTP packet and B5 last.

4.5.6 G729

 G729 is specified in ITU-T Recommendation G.729, "Coding of speech at
 8 kbit/s using conjugate structure-algebraic code excited linear
 prediction (CS-ACELP)".  A reduced-complexity version of the G.729
 algorithm is specified in Annex A to Rec. G.729.  The speech coding
 algorithms in the main body of G.729 and in G.729 Annex A are fully
 interoperable with each other, so there is no need to further
 distinguish between them.  An implementation that signals or accepts
 use of G729 payload format may implement either G.729 or G.729A
 unless restricted by additional signaling specified elsewhere related
 specifically to the encoding rather than the payload format.  The
 G.729 and G.729 Annex A codecs were optimized to represent speech
 with high quality, where G.729 Annex A trades some speech quality for
 an approximate 50% complexity reduction [10].  See the next Section
 (4.5.7) for other data rates added in later G.729 Annexes.  For all
 data rates, the sampling frequency (and RTP timestamp clock rate) is
 8,000 Hz.
 A voice activity detector (VAD) and comfort noise generator (CNG)
 algorithm in Annex B of G.729 is RECOMMENDED for digital simultaneous
 voice and data applications and can be used in conjunction with G.729
 or G.729 Annex A.  A G.729 or G.729 Annex A frame contains 10 octets,
 while the G.729 Annex B comfort noise frame occupies 2 octets.
 Receivers MUST accept comfort noise frames if restriction of their
 use has not been signaled.  The MIME registration for G729 in RFC
 3555 [7] specifies a parameter that MAY be used with MIME or SDP to
 restrict the use of comfort noise frames.
 A G729 RTP packet may consist of zero or more G.729 or G.729 Annex A
 frames, followed by zero or one G.729 Annex B frames.  The presence
 of a comfort noise frame can be deduced from the length of the RTP
 payload.  The default packetization interval is 20 ms (two frames),
 but in some situations it may be desirable to send 10 ms packets.  An

Schulzrinne & Casner Standards Track [Page 20] RFC 3551 RTP A/V Profile July 2003

 example would be a transition from speech to comfort noise in the
 first 10 ms of the packet.  For some applications, a longer
 packetization interval may be required to reduce the packet rate.
     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
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |L|      L1     |    L2   |    L3   |       P1      |P|    C1   |
    |0|             |         |         |               |0|         |
    | |0 1 2 3 4 5 6|0 1 2 3 4|0 1 2 3 4|0 1 2 3 4 5 6 7| |0 1 2 3 4|
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |       C1      |  S1   | GA1 |  GB1  |    P2   |      C2       |
    |          1 1 1|       |     |       |         |               |
    |5 6 7 8 9 0 1 2|0 1 2 3|0 1 2|0 1 2 3|0 1 2 3 4|0 1 2 3 4 5 6 7|
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |   C2    |  S2   | GA2 |  GB2  |
    |    1 1 1|       |     |       |
    |8 9 0 1 2|0 1 2 3|0 1 2|0 1 2 3|
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
                  Figure 4: G.729 and G.729A bit packing
 The transmitted parameters of a G.729/G.729A 10-ms frame, consisting
 of 80 bits, are defined in Recommendation G.729, Table 8/G.729.  The
 mapping of the these parameters is given below in Fig. 4.  The
 diagrams show the bit packing in "network byte order", also known as
 big-endian order.  The bits of each 32-bit word are numbered 0 to 31,
 with the most significant bit on the left and numbered 0.  The octets
 (bytes) of each word are transmitted most significant octet first.
 The bits of each data field are numbered in the order as produced by
 the G.729 C code reference implementation.
 The packing of the G.729 Annex B comfort noise frame is shown in Fig.
 5.
        0                   1
        0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       |L|  LSF1   |  LSF2 |   GAIN  |R|
       |S|         |       |         |E|
       |F|         |       |         |S|
       |0|0 1 2 3 4|0 1 2 3|0 1 2 3 4|V|    RESV = Reserved (zero)
       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
                     Figure 5: G.729 Annex B bit packing

Schulzrinne & Casner Standards Track [Page 21] RFC 3551 RTP A/V Profile July 2003

4.5.7 G729D and G729E

 Annexes D and E to ITU-T Recommendation G.729 provide additional data
 rates.  Because the data rate is not signaled in the bitstream, the
 different data rates are given distinct RTP encoding names which are
 mapped to distinct payload type numbers.  G729D indicates a 6.4
 kbit/s coding mode (G.729 Annex D, for momentary reduction in channel
 capacity), while G729E indicates an 11.8 kbit/s mode (G.729 Annex E,
 for improved performance with a wide range of narrow-band input
 signals, e.g., music and background noise).  Annex E has two
 operating modes, backward adaptive and forward adaptive, which are
 signaled by the first two bits in each frame (the most significant
 two bits of the first octet).
 The voice activity detector (VAD) and comfort noise generator (CNG)
 algorithm specified in Annex B of G.729 may be used with Annex D and
 Annex E frames in addition to G.729 and G.729 Annex A frames.  The
 algorithm details for the operation of Annexes D and E with the Annex
 B CNG are specified in G.729 Annexes F and G.  Note that Annexes F
 and G do not introduce any new encodings.  Receivers MUST accept
 comfort noise frames if restriction of their use has not been
 signaled.  The MIME registrations for G729D and G729E in RFC 3555 [7]
 specify a parameter that MAY be used with MIME or SDP to restrict the
 use of comfort noise frames.
 For G729D, an RTP packet may consist of zero or more G.729 Annex D
 frames, followed by zero or one G.729 Annex B frame.  Similarly, for
 G729E, an RTP packet may consist of zero or more G.729 Annex E
 frames, followed by zero or one G.729 Annex B frame.  The presence of
 a comfort noise frame can be deduced from the length of the RTP
 payload.
 A single RTP packet must contain frames of only one data rate,
 optionally followed by one comfort noise frame.  The data rate may be
 changed from packet to packet by changing the payload type number.
 G.729 Annexes D, E and H describe what the encoding and decoding
 algorithms must do to accommodate a change in data rate.
 For G729D, the bits of a G.729 Annex D frame are formatted as shown
 below in Fig. 6 (cf.  Table D.1/G.729).  The frame length is 64 bits.

Schulzrinne & Casner Standards Track [Page 22] RFC 3551 RTP A/V Profile July 2003

     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
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |L|      L1     |    L2   |    L3   |        P1     |     C1    |
    |0|             |         |         |               |           |
    | |0 1 2 3 4 5 6|0 1 2 3 4|0 1 2 3 4|0 1 2 3 4 5 6 7|0 1 2 3 4 5|
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    | C1  |S1 | GA1 | GB1 |  P2   |        C2       |S2 | GA2 | GB2 |
    |     |   |     |     |       |                 |   |     |     |
    |6 7 8|0 1|0 1 2|0 1 2|0 1 2 3|0 1 2 3 4 5 6 7 8|0 1|0 1 2|0 1 2|
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
                   Figure 6: G.729 Annex D bit packing
 The net bit rate for the G.729 Annex E algorithm is 11.8 kbit/s and a
 total of 118 bits are used.  Two bits are appended as "don't care"
 bits to complete an integer number of octets for the frame.  For
 G729E, the bits of a data frame are formatted as shown in the next
 two diagrams (cf. Table E.1/G.729).  The fields for the G729E forward
 adaptive mode are packed as shown in Fig. 7.
     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|L|      L1     |    L2   |    L3   |        P1     |P| C0_1|
    |   |0|             |         |         |               |0|     |
    |   | |0 1 2 3 4 5 6|0 1 2 3 4|0 1 2 3 4|0 1 2 3 4 5 6 7| |0 1 2|
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |       |   C1_1      |     C2_1    |   C3_1      |    C4_1     |
    |       |             |             |             |             |
    |3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    | GA1 |  GB1  |    P2   |   C0_2      |     C1_2    |   C2_2    |
    |     |       |         |             |             |           |
    |0 1 2|0 1 2 3|0 1 2 3 4|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5|
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    | |    C3_2     |     C4_2    | GA2 | GB2   |DC |
    | |             |             |     |       |   |
    |6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2|0 1 2 3|0 1|
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       Figure 7: G.729 Annex E (forward adaptive mode) bit packing
 The fields for the G729E backward adaptive mode are packed as shown
 in Fig. 8.

Schulzrinne & Casner Standards Track [Page 23] RFC 3551 RTP A/V Profile July 2003

     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 1|       P1      |P|       C0_1              |     C1_1      |
    |   |               |0|                    1 1 1|               |
    |   |0 1 2 3 4 5 6 7|0|0 1 2 3 4 5 6 7 8 9 0 1 2|0 1 2 3 4 5 6 7|
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |   |  C2_1       | C3_1        | C4_1        |GA1  | GB1   |P2 |
    |   |             |             |             |     |       |   |
    |8 9|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2|0 1 2 3|0 1|
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |     |          C0_2           |       C1_2        |    C2_2   |
    |     |                    1 1 1|                   |           |
    |2 3 4|0 1 2 3 4 5 6 7 8 9 0 1 2|0 1 2 3 4 5 6 7 8 9|0 1 2 3 4 5|
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    | |    C3_2     |     C4_2    | GA2 | GB2   |DC |
    | |             |             |     |       |   |
    |6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2|0 1 2 3|0 1|
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
       Figure 8: G.729 Annex E (backward adaptive mode) bit packing

4.5.8 GSM

 GSM (Group Speciale Mobile) denotes the European GSM 06.10 standard
 for full-rate speech transcoding, ETS 300 961, which is based on
 RPE/LTP (residual pulse excitation/long term prediction) coding at a
 rate of 13 kb/s [11,12,13].  The text of the standard can be obtained
 from:
 ETSI (European Telecommunications Standards Institute)
 ETSI Secretariat: B.P.152
 F-06561 Valbonne Cedex
 France
 Phone: +33 92 94 42 00
 Fax:   +33 93 65 47 16
 Blocks of 160 audio samples are compressed into 33 octets, for an
 effective data rate of 13,200 b/s.

4.5.8.1 General Packaging Issues

 The GSM standard (ETS 300 961) specifies the bit stream produced by
 the codec, but does not specify how these bits should be packed for
 transmission.  The packetization specified here has subsequently been
 adopted in ETSI Technical Specification TS 101 318.  Some software
 implementations of the GSM codec use a different packing than that
 specified here.

Schulzrinne & Casner Standards Track [Page 24] RFC 3551 RTP A/V Profile July 2003

             field  field name  bits  field  field name  bits
             ________________________________________________
             1      LARc[0]     6     39     xmc[22]     3
             2      LARc[1]     6     40     xmc[23]     3
             3      LARc[2]     5     41     xmc[24]     3
             4      LARc[3]     5     42     xmc[25]     3
             5      LARc[4]     4     43     Nc[2]       7
             6      LARc[5]     4     44     bc[2]       2
             7      LARc[6]     3     45     Mc[2]       2
             8      LARc[7]     3     46     xmaxc[2]    6
             9      Nc[0]       7     47     xmc[26]     3
             10     bc[0]       2     48     xmc[27]     3
             11     Mc[0]       2     49     xmc[28]     3
             12     xmaxc[0]    6     50     xmc[29]     3
             13     xmc[0]      3     51     xmc[30]     3
             14     xmc[1]      3     52     xmc[31]     3
             15     xmc[2]      3     53     xmc[32]     3
             16     xmc[3]      3     54     xmc[33]     3
             17     xmc[4]      3     55     xmc[34]     3
             18     xmc[5]      3     56     xmc[35]     3
             19     xmc[6]      3     57     xmc[36]     3
             20     xmc[7]      3     58     xmc[37]     3
             21     xmc[8]      3     59     xmc[38]     3
             22     xmc[9]      3     60     Nc[3]       7
             23     xmc[10]     3     61     bc[3]       2
             24     xmc[11]     3     62     Mc[3]       2
             25     xmc[12]     3     63     xmaxc[3]    6
             26     Nc[1]       7     64     xmc[39]     3
             27     bc[1]       2     65     xmc[40]     3
             28     Mc[1]       2     66     xmc[41]     3
             29     xmaxc[1]    6     67     xmc[42]     3
             30     xmc[13]     3     68     xmc[43]     3
             31     xmc[14]     3     69     xmc[44]     3
             32     xmc[15]     3     70     xmc[45]     3
             33     xmc[16]     3     71     xmc[46]     3
             34     xmc[17]     3     72     xmc[47]     3
             35     xmc[18]     3     73     xmc[48]     3
             36     xmc[19]     3     74     xmc[49]     3
             37     xmc[20]     3     75     xmc[50]     3
             38     xmc[21]     3     76     xmc[51]     3
                    Table 2: Ordering of GSM variables

Schulzrinne & Casner Standards Track [Page 25] RFC 3551 RTP A/V Profile July 2003

 Octet  Bit 0   Bit 1   Bit 2   Bit 3   Bit 4   Bit 5   Bit 6   Bit 7
 _____________________________________________________________________
     0    1       1       0       1    LARc0.0 LARc0.1 LARc0.2 LARc0.3
     1 LARc0.4 LARc0.5 LARc1.0 LARc1.1 LARc1.2 LARc1.3 LARc1.4 LARc1.5
     2 LARc2.0 LARc2.1 LARc2.2 LARc2.3 LARc2.4 LARc3.0 LARc3.1 LARc3.2
     3 LARc3.3 LARc3.4 LARc4.0 LARc4.1 LARc4.2 LARc4.3 LARc5.0 LARc5.1
     4 LARc5.2 LARc5.3 LARc6.0 LARc6.1 LARc6.2 LARc7.0 LARc7.1 LARc7.2
     5  Nc0.0   Nc0.1   Nc0.2   Nc0.3   Nc0.4   Nc0.5   Nc0.6  bc0.0
     6  bc0.1   Mc0.0   Mc0.1  xmaxc00 xmaxc01 xmaxc02 xmaxc03 xmaxc04
     7 xmaxc05 xmc0.0  xmc0.1  xmc0.2  xmc1.0  xmc1.1  xmc1.2  xmc2.0
     8 xmc2.1  xmc2.2  xmc3.0  xmc3.1  xmc3.2  xmc4.0  xmc4.1  xmc4.2
     9 xmc5.0  xmc5.1  xmc5.2  xmc6.0  xmc6.1  xmc6.2  xmc7.0  xmc7.1
    10 xmc7.2  xmc8.0  xmc8.1  xmc8.2  xmc9.0  xmc9.1  xmc9.2  xmc10.0
    11 xmc10.1 xmc10.2 xmc11.0 xmc11.1 xmc11.2 xmc12.0 xmc12.1 xcm12.2
    12  Nc1.0   Nc1.1   Nc1.2   Nc1.3   Nc1.4   Nc1.5   Nc1.6   bc1.0
    13  bc1.1   Mc1.0   Mc1.1  xmaxc10 xmaxc11 xmaxc12 xmaxc13 xmaxc14
    14 xmax15  xmc13.0 xmc13.1 xmc13.2 xmc14.0 xmc14.1 xmc14.2 xmc15.0
    15 xmc15.1 xmc15.2 xmc16.0 xmc16.1 xmc16.2 xmc17.0 xmc17.1 xmc17.2
    16 xmc18.0 xmc18.1 xmc18.2 xmc19.0 xmc19.1 xmc19.2 xmc20.0 xmc20.1
    17 xmc20.2 xmc21.0 xmc21.1 xmc21.2 xmc22.0 xmc22.1 xmc22.2 xmc23.0
    18 xmc23.1 xmc23.2 xmc24.0 xmc24.1 xmc24.2 xmc25.0 xmc25.1 xmc25.2
    19  Nc2.0   Nc2.1   Nc2.2   Nc2.3   Nc2.4   Nc2.5   Nc2.6   bc2.0
    20  bc2.1   Mc2.0   Mc2.1  xmaxc20 xmaxc21 xmaxc22 xmaxc23 xmaxc24
    21 xmaxc25 xmc26.0 xmc26.1 xmc26.2 xmc27.0 xmc27.1 xmc27.2 xmc28.0
    22 xmc28.1 xmc28.2 xmc29.0 xmc29.1 xmc29.2 xmc30.0 xmc30.1 xmc30.2
    23 xmc31.0 xmc31.1 xmc31.2 xmc32.0 xmc32.1 xmc32.2 xmc33.0 xmc33.1
    24 xmc33.2 xmc34.0 xmc34.1 xmc34.2 xmc35.0 xmc35.1 xmc35.2 xmc36.0
    25 Xmc36.1 xmc36.2 xmc37.0 xmc37.1 xmc37.2 xmc38.0 xmc38.1 xmc38.2
    26  Nc3.0   Nc3.1   Nc3.2   Nc3.3   Nc3.4   Nc3.5   Nc3.6   bc3.0
    27  bc3.1   Mc3.0   Mc3.1  xmaxc30 xmaxc31 xmaxc32 xmaxc33 xmaxc34
    28 xmaxc35 xmc39.0 xmc39.1 xmc39.2 xmc40.0 xmc40.1 xmc40.2 xmc41.0
    29 xmc41.1 xmc41.2 xmc42.0 xmc42.1 xmc42.2 xmc43.0 xmc43.1 xmc43.2
    30 xmc44.0 xmc44.1 xmc44.2 xmc45.0 xmc45.1 xmc45.2 xmc46.0 xmc46.1
    31 xmc46.2 xmc47.0 xmc47.1 xmc47.2 xmc48.0 xmc48.1 xmc48.2 xmc49.0
    32 xmc49.1 xmc49.2 xmc50.0 xmc50.1 xmc50.2 xmc51.0 xmc51.1 xmc51.2
                      Table 3: GSM payload format
 In the GSM packing used by RTP, the bits SHALL be packed beginning
 from the most significant bit.  Every 160 sample GSM frame is coded
 into one 33 octet (264 bit) buffer.  Every such buffer begins with a
 4 bit signature (0xD), followed by the MSB encoding of the fields of
 the frame.  The first octet thus contains 1101 in the 4 most
 significant bits (0-3) and the 4 most significant bits of F1 (0-3) in
 the 4 least significant bits (4-7).  The second octet contains the 2
 least significant bits of F1 in bits 0-1, and F2 in bits 2-7, and so
 on.  The order of the fields in the frame is described in Table 2.

Schulzrinne & Casner Standards Track [Page 26] RFC 3551 RTP A/V Profile July 2003

4.5.8.2 GSM Variable Names and Numbers

 In the RTP encoding we have the bit pattern described in Table 3,
 where F.i signifies the ith bit of the field F, bit 0 is the most
 significant bit, and the bits of every octet are numbered from 0 to 7
 from most to least significant.

4.5.9 GSM-EFR

 GSM-EFR denotes GSM 06.60 enhanced full rate speech transcoding,
 specified in ETS 300 726 which is available from ETSI at the address
 given in Section 4.5.8.  This codec has a frame length of 244 bits.
 For transmission in RTP, each codec frame is packed into a 31 octet
 (248 bit) buffer beginning with a 4-bit signature 0xC in a manner
 similar to that specified here for the original GSM 06.10 codec.  The
 packing is specified in ETSI Technical Specification TS 101 318.

4.5.10 L8

 L8 denotes linear audio data samples, using 8-bits of precision with
 an offset of 128, that is, the most negative signal is encoded as
 zero.

4.5.11 L16

 L16 denotes uncompressed audio data samples, using 16-bit signed
 representation with 65,535 equally divided steps between minimum and
 maximum signal level, ranging from -32,768 to 32,767.  The value is
 represented in two's complement notation and transmitted in network
 byte order (most significant byte first).
 The MIME registration for L16 in RFC 3555 [7] specifies parameters
 that MAY be used with MIME or SDP to indicate that analog pre-
 emphasis was applied to the signal before quantization or to indicate
 that a multiple-channel audio stream follows a different channel
 ordering convention than is specified in Section 4.1.

4.5.12 LPC

 LPC designates an experimental linear predictive encoding contributed
 by Ron Frederick, which is based on an implementation written by Ron
 Zuckerman posted to the Usenet group comp.dsp on June 26, 1992.  The
 codec generates 14 octets for every frame.  The framesize is set to
 20 ms, resulting in a bit rate of 5,600 b/s.

Schulzrinne & Casner Standards Track [Page 27] RFC 3551 RTP A/V Profile July 2003

4.5.13 MPA

 MPA denotes MPEG-1 or MPEG-2 audio encapsulated as elementary
 streams.  The encoding is defined in ISO standards ISO/IEC 11172-3
 and 13818-3.  The encapsulation is specified in RFC 2250 [14].
 The encoding may be at any of three levels of complexity, called
 Layer I, II and III.  The selected layer as well as the sampling rate
 and channel count are indicated in the payload.  The RTP timestamp
 clock rate is always 90,000, independent of the sampling rate.
 MPEG-1 audio supports sampling rates of 32, 44.1, and 48 kHz (ISO/IEC
 11172-3, section 1.1; "Scope").  MPEG-2 supports sampling rates of
 16, 22.05 and 24 kHz.  The number of samples per frame is fixed, but
 the frame size will vary with the sampling rate and bit rate.
 The MIME registration for MPA in RFC 3555 [7] specifies parameters
 that MAY be used with MIME or SDP to restrict the selection of layer,
 channel count, sampling rate, and bit rate.

4.5.14 PCMA and PCMU

 PCMA and PCMU are specified in ITU-T Recommendation G.711.  Audio
 data is encoded as eight bits per sample, after logarithmic scaling.
 PCMU denotes mu-law scaling, PCMA A-law scaling.  A detailed
 description is given by Jayant and Noll [15].  Each G.711 octet SHALL
 be octet-aligned in an RTP packet.  The sign bit of each G.711 octet
 SHALL correspond to the most significant bit of the octet in the RTP
 packet (i.e., assuming the G.711 samples are handled as octets on the
 host machine, the sign bit SHALL be the most significant bit of the
 octet as defined by the host machine format).  The 56 kb/s and 48
 kb/s modes of G.711 are not applicable to RTP, since PCMA and PCMU
 MUST always be transmitted as 8-bit samples.
 See Section 4.1 regarding silence suppression.

4.5.15 QCELP

 The Electronic Industries Association (EIA) & Telecommunications
 Industry Association (TIA) standard IS-733, "TR45: High Rate Speech
 Service Option for Wideband Spread Spectrum Communications Systems",
 defines the QCELP audio compression algorithm for use in wireless
 CDMA applications.  The QCELP CODEC compresses each 20 milliseconds
 of 8,000 Hz, 16-bit sampled input speech into one of four different
 size output frames:  Rate 1 (266 bits), Rate 1/2 (124 bits), Rate 1/4
 (54 bits) or Rate 1/8 (20 bits).  For typical speech patterns, this
 results in an average output of 6.8 kb/s for normal mode and 4.7 kb/s
 for reduced rate mode.  The packetization of the QCELP audio codec is
 described in [16].

Schulzrinne & Casner Standards Track [Page 28] RFC 3551 RTP A/V Profile July 2003

4.5.16 RED

 The redundant audio payload format "RED" is specified by RFC 2198
 [17].  It defines a means by which multiple redundant copies of an
 audio packet may be transmitted in a single RTP stream.  Each packet
 in such a stream contains, in addition to the audio data for that
 packetization interval, a (more heavily compressed) copy of the data
 from a previous packetization interval.  This allows an approximation
 of the data from lost packets to be recovered upon decoding of a
 subsequent packet, giving much improved sound quality when compared
 with silence substitution for lost packets.

4.5.17 VDVI

 VDVI is a variable-rate version of DVI4, yielding speech bit rates of
 between 10 and 25 kb/s.  It is specified for single-channel operation
 only.  Samples are packed into octets starting at the most-
 significant bit.  The last octet is padded with 1 bits if the last
 sample does not fill the last octet.  This padding is distinct from
 the valid codewords.  The receiver needs to detect the padding
 because there is no explicit count of samples in the packet.
 It uses the following encoding:
          DVI4 codeword  VDVI bit pattern
          _______________________________
                      0  00
                      1  010
                      2  1100
                      3  11100
                      4  111100
                      5  1111100
                      6  11111100
                      7  11111110
                      8  10
                      9  011
                     10  1101
                     11  11101
                     12  111101
                     13  1111101
                     14  11111101
                     15  11111111

Schulzrinne & Casner Standards Track [Page 29] RFC 3551 RTP A/V Profile July 2003

5. Video

 The following sections describe the video encodings that are defined
 in this memo and give their abbreviated names used for
 identification.  These video encodings and their payload types are
 listed in Table 5.
 All of these video encodings use an RTP timestamp frequency of 90,000
 Hz, the same as the MPEG presentation time stamp frequency.  This
 frequency yields exact integer timestamp increments for the typical
 24 (HDTV), 25 (PAL), and 29.97 (NTSC) and 30 Hz (HDTV) frame rates
 and 50, 59.94 and 60 Hz field rates.  While 90 kHz is the RECOMMENDED
 rate for future video encodings used within this profile, other rates
 MAY be used.  However, it is not sufficient to use the video frame
 rate (typically between 15 and 30 Hz) because that does not provide
 adequate resolution for typical synchronization requirements when
 calculating the RTP timestamp corresponding to the NTP timestamp in
 an RTCP SR packet.  The timestamp resolution MUST also be sufficient
 for the jitter estimate contained in the receiver reports.
 For most of these video encodings, the RTP timestamp encodes the
 sampling instant of the video image contained in the RTP data packet.
 If a video image occupies more than one packet, the timestamp is the
 same on all of those packets.  Packets from different video images
 are distinguished by their different timestamps.
 Most of these video encodings also specify that the marker bit of the
 RTP header SHOULD be set to one in the last packet of a video frame
 and otherwise set to zero.  Thus, it is not necessary to wait for a
 following packet with a different timestamp to detect that a new
 frame should be displayed.

5.1 CelB

 The CELL-B encoding is a proprietary encoding proposed by Sun
 Microsystems.  The byte stream format is described in RFC 2029 [18].

5.2 JPEG

 The encoding is specified in ISO Standards 10918-1 and 10918-2.  The
 RTP payload format is as specified in RFC 2435 [19].

5.3 H261

 The encoding is specified in ITU-T Recommendation H.261, "Video codec
 for audiovisual services at p x 64 kbit/s".  The packetization and
 RTP-specific properties are described in RFC 2032 [20].

Schulzrinne & Casner Standards Track [Page 30] RFC 3551 RTP A/V Profile July 2003

5.4 H263

 The encoding is specified in the 1996 version of ITU-T Recommendation
 H.263, "Video coding for low bit rate communication".  The
 packetization and RTP-specific properties are described in RFC 2190
 [21].  The H263-1998 payload format is RECOMMENDED over this one for
 use by new implementations.

5.5 H263-1998

 The encoding is specified in the 1998 version of ITU-T Recommendation
 H.263, "Video coding for low bit rate communication".  The
 packetization and RTP-specific properties are described in RFC 2429
 [22].  Because the 1998 version of H.263 is a superset of the 1996
 syntax, this payload format can also be used with the 1996 version of
 H.263, and is RECOMMENDED for this use by new implementations.  This
 payload format does not replace RFC 2190, which continues to be used
 by existing implementations, and may be required for backward
 compatibility in new implementations.  Implementations using the new
 features of the 1998 version of H.263 MUST use the payload format
 described in RFC 2429.

5.6 MPV

 MPV designates the use of MPEG-1 and MPEG-2 video encoding elementary
 streams as specified in ISO Standards ISO/IEC 11172 and 13818-2,
 respectively.  The RTP payload format is as specified in RFC 2250
 [14], Section 3.
 The MIME registration for MPV in RFC 3555 [7] specifies a parameter
 that MAY be used with MIME or SDP to restrict the selection of the
 type of MPEG video.

5.7 MP2T

 MP2T designates the use of MPEG-2 transport streams, for either audio
 or video.  The RTP payload format is described in RFC 2250 [14],
 Section 2.

Schulzrinne & Casner Standards Track [Page 31] RFC 3551 RTP A/V Profile July 2003

5.8 nv

 The encoding is implemented in the program `nv', version 4, developed
 at Xerox PARC by Ron Frederick.  Further information is available
 from the author:
 Ron Frederick
 Blue Coat Systems Inc.
 650 Almanor Avenue
 Sunnyvale, CA 94085
 United States
 EMail: ronf@bluecoat.com

6. Payload Type Definitions

 Tables 4 and 5 define this profile's static payload type values for
 the PT field of the RTP data header.  In addition, payload type
 values in the range 96-127 MAY be defined dynamically through a
 conference control protocol, which is beyond the scope of this
 document.  For example, a session directory could specify that for a
 given session, payload type 96 indicates PCMU encoding, 8,000 Hz
 sampling rate, 2 channels.  Entries in Tables 4 and 5 with payload
 type "dyn" have no static payload type assigned and are only used
 with a dynamic payload type.  Payload type 2 was assigned to G721 in
 RFC 1890 and to its equivalent successor G726-32 in draft versions of
 this specification, but its use is now deprecated and that static
 payload type is marked reserved due to conflicting use for the
 payload formats G726-32 and AAL2-G726-32 (see Section 4.5.4).
 Payload type 13 indicates the Comfort Noise (CN) payload format
 specified in RFC 3389 [9].  Payload type 19 is marked "reserved"
 because some draft versions of this specification assigned that
 number to an earlier version of the comfort noise payload format.
 The payload type range 72-76 is marked "reserved" so that RTCP and
 RTP packets can be reliably distinguished (see Section "Summary of
 Protocol Constants" of the RTP protocol specification).
 The payload types currently defined in this profile are assigned to
 exactly one of three categories or media types:  audio only, video
 only and those combining audio and video.  The media types are marked
 in Tables 4 and 5 as "A", "V" and "AV", respectively.  Payload types
 of different media types SHALL NOT be interleaved or multiplexed
 within a single RTP session, but multiple RTP sessions MAY be used in
 parallel to send multiple media types.  An RTP source MAY change
 payload types within the same media type during a session.  See the
 section "Multiplexing RTP Sessions" of RFC 3550 for additional
 explanation.

Schulzrinne & Casner Standards Track [Page 32] RFC 3551 RTP A/V Profile July 2003

             PT   encoding    media type  clock rate   channels
                  name                    (Hz)
             ___________________________________________________
             0    PCMU        A            8,000       1
             1    reserved    A
             2    reserved    A
             3    GSM         A            8,000       1
             4    G723        A            8,000       1
             5    DVI4        A            8,000       1
             6    DVI4        A           16,000       1
             7    LPC         A            8,000       1
             8    PCMA        A            8,000       1
             9    G722        A            8,000       1
             10   L16         A           44,100       2
             11   L16         A           44,100       1
             12   QCELP       A            8,000       1
             13   CN          A            8,000       1
             14   MPA         A           90,000       (see text)
             15   G728        A            8,000       1
             16   DVI4        A           11,025       1
             17   DVI4        A           22,050       1
             18   G729        A            8,000       1
             19   reserved    A
             20   unassigned  A
             21   unassigned  A
             22   unassigned  A
             23   unassigned  A
             dyn  G726-40     A            8,000       1
             dyn  G726-32     A            8,000       1
             dyn  G726-24     A            8,000       1
             dyn  G726-16     A            8,000       1
             dyn  G729D       A            8,000       1
             dyn  G729E       A            8,000       1
             dyn  GSM-EFR     A            8,000       1
             dyn  L8          A            var.        var.
             dyn  RED         A                        (see text)
             dyn  VDVI        A            var.        1
             Table 4: Payload types (PT) for audio encodings

Schulzrinne & Casner Standards Track [Page 33] RFC 3551 RTP A/V Profile July 2003

             PT      encoding    media type  clock rate
                     name                    (Hz)
             _____________________________________________
             24      unassigned  V
             25      CelB        V           90,000
             26      JPEG        V           90,000
             27      unassigned  V
             28      nv          V           90,000
             29      unassigned  V
             30      unassigned  V
             31      H261        V           90,000
             32      MPV         V           90,000
             33      MP2T        AV          90,000
             34      H263        V           90,000
             35-71   unassigned  ?
             72-76   reserved    N/A         N/A
             77-95   unassigned  ?
             96-127  dynamic     ?
             dyn     H263-1998   V           90,000
             Table 5: Payload types (PT) for video and combined
                      encodings
 Session participants agree through mechanisms beyond the scope of
 this specification on the set of payload types allowed in a given
 session.  This set MAY, for example, be defined by the capabilities
 of the applications used, negotiated by a conference control protocol
 or established by agreement between the human participants.
 Audio applications operating under this profile SHOULD, at a minimum,
 be able to send and/or receive payload types 0 (PCMU) and 5 (DVI4).
 This allows interoperability without format negotiation and ensures
 successful negotiation with a conference control protocol.

7. RTP over TCP and Similar Byte Stream Protocols

 Under special circumstances, it may be necessary to carry RTP in
 protocols offering a byte stream abstraction, such as TCP, possibly
 multiplexed with other data.  The application MUST define its own
 method of delineating RTP and RTCP packets (RTSP [23] provides an
 example of such an encapsulation specification).

8. Port Assignment

 As specified in the RTP protocol definition, RTP data SHOULD be
 carried on an even UDP port number and the corresponding RTCP packets
 SHOULD be carried on the next higher (odd) port number.

Schulzrinne & Casner Standards Track [Page 34] RFC 3551 RTP A/V Profile July 2003

 Applications operating under this profile MAY use any such UDP port
 pair.  For example, the port pair MAY be allocated randomly by a
 session management program.  A single fixed port number pair cannot
 be required because multiple applications using this profile are
 likely to run on the same host, and there are some operating systems
 that do not allow multiple processes to use the same UDP port with
 different multicast addresses.
 However, port numbers 5004 and 5005 have been registered for use with
 this profile for those applications that choose to use them as the
 default pair.  Applications that operate under multiple profiles MAY
 use this port pair as an indication to select this profile if they
 are not subject to the constraint of the previous paragraph.
 Applications need not have a default and MAY require that the port
 pair be explicitly specified.  The particular port numbers were
 chosen to lie in the range above 5000 to accommodate port number
 allocation practice within some versions of the Unix operating
 system, where port numbers below 1024 can only be used by privileged
 processes and port numbers between 1024 and 5000 are automatically
 assigned by the operating system.

9. Changes from RFC 1890

 This RFC revises RFC 1890.  It is mostly backwards-compatible with
 RFC 1890 except for functions removed because two interoperable
 implementations were not found.  The additions to RFC 1890 codify
 existing practice in the use of payload formats under this profile.
 Since this profile may be used without using any of the payload
 formats listed here, the addition of new payload formats in this
 revision does not affect backwards compatibility.  The changes are
 listed below, categorized into functional and non-functional changes.
 Functional changes:
 o  Section 11, "IANA Considerations" was added to specify the
    registration of the name for this profile.  That appendix also
    references a new Section 3 "Registering Additional Encodings"
    which establishes a policy that no additional registration of
    static payload types for this profile will be made beyond those
    added in this revision and included in Tables 4 and 5.  Instead,
    additional encoding names may be registered as MIME subtypes for
    binding to dynamic payload types.  Non-normative references were
    added to RFC 3555 [7] where MIME subtypes for all the listed
    payload formats are registered, some with optional parameters for
    use of the payload formats.

Schulzrinne & Casner Standards Track [Page 35] RFC 3551 RTP A/V Profile July 2003

 o  Static payload types 4, 16, 17 and 34 were added to incorporate
    IANA registrations made since the publication of RFC 1890, along
    with the corresponding payload format descriptions for G723 and
    H263.
 o  Following working group discussion, static payload types 12 and 18
    were added along with the corresponding payload format
    descriptions for QCELP and G729.  Static payload type 13 was
    assigned to the Comfort Noise (CN) payload format defined in RFC
    3389.  Payload type 19 was marked reserved because it had been
    temporarily allocated to an earlier version of Comfort Noise
    present in some draft revisions of this document.
 o  The payload format for G721 was renamed to G726-32 following the
    ITU-T renumbering, and the payload format description for G726 was
    expanded to include the -16, -24 and -40 data rates.  Because of
    confusion regarding draft revisions of this document, some
    implementations of these G726 payload formats packed samples into
    octets starting with the most significant bit rather than the
    least significant bit as specified here.  To partially resolve
    this incompatibility, new payload formats named AAL2-G726-16, -24,
    -32 and -40 will be specified in a separate document (see note in
    Section 4.5.4), and use of static payload type 2 is deprecated as
    explained in Section 6.
 o  Payload formats G729D and G729E were added following the ITU-T
    addition of Annexes D and E to Recommendation G.729.  Listings
    were added for payload formats GSM-EFR, RED, and H263-1998
    published in other documents subsequent to RFC 1890.  These
    additional payload formats are referenced only by dynamic payload
    type numbers.
 o  The descriptions of the payload formats for G722, G728, GSM, VDVI
    were expanded.
 o  The payload format for 1016 audio was removed and its static
    payload type assignment 1 was marked "reserved" because two
    interoperable implementations were not found.
 o  Requirements for congestion control were added in Section 2.
 o  This profile follows the suggestion in the revised RTP spec that
    RTCP bandwidth may be specified separately from the session
    bandwidth and separately for active senders and passive receivers.
 o  The mapping of a user pass-phrase string into an encryption key
    was deleted from Section 2 because two interoperable
    implementations were not found.

Schulzrinne & Casner Standards Track [Page 36] RFC 3551 RTP A/V Profile July 2003

 o  The "quadrophonic" sample ordering convention for four-channel
    audio was removed to eliminate an ambiguity as noted in Section
    4.1.
 Non-functional changes:
 o  In Section 4.1, it is now explicitly stated that silence
    suppression is allowed for all audio payload formats.  (This has
    always been the case and derives from a fundamental aspect of
    RTP's design and the motivations for packet audio, but was not
    explicit stated before.)  The use of comfort noise is also
    explained.
 o  In Section 4.1, the requirement level for setting of the marker
    bit on the first packet after silence for audio was changed from
    "is" to "SHOULD be", and clarified that the marker bit is set only
    when packets are intentionally not sent.
 o  Similarly, text was added to specify that the marker bit SHOULD be
    set to one on the last packet of a video frame, and that video
    frames are distinguished by their timestamps.
 o  RFC references are added for payload formats published after RFC
    1890.
 o  The security considerations and full copyright sections were
    added.
 o  According to Peter Hoddie of Apple, only pre-1994 Macintosh used
    the 22254.54 rate and none the 11127.27 rate, so the latter was
    dropped from the discussion of suggested sampling frequencies.
 o  Table 1 was corrected to move some values from the "ms/packet"
    column to the "default ms/packet" column where they belonged.
 o  Since the Interactive Multimedia Association ceased operations, an
    alternate resource was provided for a referenced IMA document.
 o  A note has been added for G722 to clarify a discrepancy between
    the actual sampling rate and the RTP timestamp clock rate.
 o  Small clarifications of the text have been made in several places,
    some in response to questions from readers.  In particular:
  1. A definition for "media type" is given in Section 1.1 to allow

the explanation of multiplexing RTP sessions in Section 6 to be

       more clear regarding the multiplexing of multiple media.

Schulzrinne & Casner Standards Track [Page 37] RFC 3551 RTP A/V Profile July 2003

  1. The explanation of how to determine the number of audio frames

in a packet from the length was expanded.

  1. More description of the allocation of bandwidth to SDES items

is given.

  1. A note was added that the convention for the order of channels

specified in Section 4.1 may be overridden by a particular

       encoding or payload format specification.
  1. The terms MUST, SHOULD, MAY, etc. are used as defined in RFC

2119.

 o  A second author for this document was added.

10. Security Considerations

 Implementations using the profile defined in this specification are
 subject to the security considerations discussed in the RTP
 specification [1].  This profile does not specify any different
 security services.  The primary function of this profile is to list a
 set of data compression encodings for audio and video media.
 Confidentiality of the media streams is achieved by encryption.
 Because the data compression used with the payload formats described
 in this profile is applied end-to-end, encryption may be performed
 after compression so there is no conflict between the two operations.
 A potential denial-of-service threat exists for data encodings using
 compression techniques that have non-uniform receiver-end
 computational load.  The attacker can inject pathological datagrams
 into the stream which are complex to decode and cause the receiver to
 be overloaded.
 As with any IP-based protocol, in some circumstances a receiver may
 be overloaded simply by the receipt of too many packets, either
 desired or undesired.  Network-layer authentication MAY be used to
 discard packets from undesired sources, but the processing cost of
 the authentication itself may be too high.  In a multicast
 environment, source pruning is implemented in IGMPv3 (RFC 3376) [24]
 and in multicast routing protocols to allow a receiver to select
 which sources are allowed to reach it.

Schulzrinne & Casner Standards Track [Page 38] RFC 3551 RTP A/V Profile July 2003

11. IANA Considerations

 The RTP specification establishes a registry of profile names for use
 by higher-level control protocols, such as the Session Description
 Protocol (SDP), RFC 2327 [6], to refer to transport methods.  This
 profile registers the name "RTP/AVP".
 Section 3 establishes the policy that no additional registration of
 static RTP payload types for this profile will be made beyond those
 added in this document revision and included in Tables 4 and 5.  IANA
 may reference that section in declining to accept any additional
 registration requests.  In Tables 4 and 5, note that types 1 and 2
 have been marked reserved and the set of "dyn" payload types included
 has been updated.  These changes are explained in Sections 6 and 9.

12. References

12.1 Normative References

 [1]  Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,
      "RTP:  A Transport Protocol for Real-Time Applications", RFC
      3550, July 2003.
 [2]  Bradner, S., "Key Words for Use in RFCs to Indicate Requirement
      Levels", BCP 14, RFC 2119, March 1997.
 [3]  Apple Computer, "Audio Interchange File Format AIFF-C", August
      1991.  (also ftp://ftp.sgi.com/sgi/aiff-c.9.26.91.ps.Z).

12.2 Informative References

 [4]  Braden, R., Clark, D. and S. Shenker, "Integrated Services in
      the Internet Architecture: an Overview", RFC 1633, June 1994.
 [5]  Blake, S., Black, D., Carlson, M., Davies, E., Wang, Z. and W.
      Weiss, "An Architecture for Differentiated Service", RFC 2475,
      December 1998.
 [6]  Handley, M. and V. Jacobson, "SDP: Session Description
      Protocol", RFC 2327, April 1998.
 [7]  Casner, S. and P. Hoschka, "MIME Type Registration of RTP
      Payload Types", RFC 3555, July 2003.
 [8]  Freed, N., Klensin, J. and J. Postel, "Multipurpose Internet
      Mail Extensions (MIME) Part Four: Registration Procedures", BCP
      13, RFC 2048, November 1996.

Schulzrinne & Casner Standards Track [Page 39] RFC 3551 RTP A/V Profile July 2003

 [9]  Zopf, R., "Real-time Transport Protocol (RTP) Payload for
      Comfort Noise (CN)", RFC 3389, September 2002.
 [10] Deleam, D. and J.-P. Petit, "Real-time implementations of the
      recent ITU-T low bit rate speech coders on the TI TMS320C54X
      DSP: results, methodology, and applications", in Proc. of
      International Conference on Signal Processing, Technology, and
      Applications (ICSPAT) , (Boston, Massachusetts), pp. 1656--1660,
      October 1996.
 [11] Mouly, M. and M.-B. Pautet, The GSM system for mobile
      communications Lassay-les-Chateaux, France: Europe Media
      Duplication, 1993.
 [12] Degener, J., "Digital Speech Compression", Dr. Dobb's Journal,
      December 1994.
 [13] Redl, S., Weber, M. and M. Oliphant, An Introduction to GSM
      Boston: Artech House, 1995.
 [14] Hoffman, D., Fernando, G., Goyal, V. and M. Civanlar, "RTP
      Payload Format for MPEG1/MPEG2 Video", RFC 2250, January 1998.
 [15] Jayant, N. and P. Noll, Digital Coding of Waveforms--Principles
      and Applications to Speech and Video Englewood Cliffs, New
      Jersey: Prentice-Hall, 1984.
 [16] McKay, K., "RTP Payload Format for PureVoice(tm) Audio", RFC
      2658, August 1999.
 [17] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., Handley, M.,
      Bolot, J.-C., Vega-Garcia, A. and S. Fosse-Parisis, "RTP Payload
      for Redundant Audio Data", RFC 2198, September 1997.
 [18] Speer, M. and D. Hoffman, "RTP Payload Format of Sun's CellB
      Video Encoding", RFC 2029, October 1996.
 [19] Berc, L., Fenner, W., Frederick, R., McCanne, S. and P. Stewart,
      "RTP Payload Format for JPEG-Compressed Video", RFC 2435,
      October 1998.
 [20] Turletti, T. and C. Huitema, "RTP Payload Format for H.261 Video
      Streams", RFC 2032, October 1996.
 [21] Zhu, C., "RTP Payload Format for H.263 Video Streams", RFC 2190,
      September 1997.

Schulzrinne & Casner Standards Track [Page 40] RFC 3551 RTP A/V Profile July 2003

 [22] Bormann, C., Cline, L., Deisher, G., Gardos, T., Maciocco, C.,
      Newell, D., Ott, J., Sullivan, G., Wenger, S. and C. Zhu, "RTP
      Payload Format for the 1998 Version of ITU-T Rec. H.263 Video
      (H.263+)", RFC 2429, October 1998.
 [23] Schulzrinne, H., Rao, A. and R. Lanphier, "Real Time Streaming
      Protocol (RTSP)", RFC 2326, April 1998.
 [24] Cain, B., Deering, S., Kouvelas, I., Fenner, B. and A.
      Thyagarajan, "Internet Group Management Protocol, Version 3",
      RFC 3376, October 2002.

13. Current Locations of Related Resources

 Note:  Several sections below refer to the ITU-T Software Tool
 Library (STL).  It is available from the ITU Sales Service, Place des
 Nations, CH-1211 Geneve 20, Switzerland (also check
 http://www.itu.int).  The ITU-T STL is covered by a license defined
 in ITU-T Recommendation G.191, "Software tools for speech and audio
 coding standardization".
 DVI4
 An archived copy of the document IMA Recommended Practices for
 Enhancing Digital Audio Compatibility in Multimedia Systems (version
 3.0), which describes the IMA ADPCM algorithm, is available at:
    http://www.cs.columbia.edu/~hgs/audio/dvi/
 An implementation is available from Jack Jansen at
    ftp://ftp.cwi.nl/local/pub/audio/adpcm.shar
 G722
 An implementation of the G.722 algorithm is available as part of the
 ITU-T STL, described above.
 G723
 The reference C code implementation defining the G.723.1 algorithm
 and its Annexes A, B, and C are available as an integral part of
 Recommendation G.723.1 from the ITU Sales Service, address listed
 above.  Both the algorithm and C code are covered by a specific
 license.  The ITU-T Secretariat should be contacted to obtain such
 licensing information.

Schulzrinne & Casner Standards Track [Page 41] RFC 3551 RTP A/V Profile July 2003

 G726
 G726 is specified in the ITU-T Recommendation G.726, "40, 32, 24, and
 16 kb/s Adaptive Differential Pulse Code Modulation (ADPCM)".  An
 implementation of the G.726 algorithm is available as part of the
 ITU-T STL, described above.
 G729
 The reference C code implementation defining the G.729 algorithm and
 its Annexes A through I are available as an integral part of
 Recommendation G.729 from the ITU Sales Service, listed above.  Annex
 I contains the integrated C source code for all G.729 operating
 modes.  The G.729 algorithm and associated C code are covered by a
 specific license.  The contact information for obtaining the license
 is available from the ITU-T Secretariat.
 GSM
 A reference implementation was written by Carsten Bormann and Jutta
 Degener (then at TU Berlin, Germany).  It is available at
    http://www.dmn.tzi.org/software/gsm/
 Although the RPE-LTP algorithm is not an ITU-T standard, there is a C
 code implementation of the RPE-LTP algorithm available as part of the
 ITU-T STL.  The STL implementation is an adaptation of the TU Berlin
 version.
 LPC
 An implementation is available at
    ftp://parcftp.xerox.com/pub/net-research/lpc.tar.Z
 PCMU, PCMA
 An implementation of these algorithms is available as part of the
 ITU-T STL, described above.

14. Acknowledgments

 The comments and careful review of Simao Campos, Richard Cox and AVT
 Working Group participants are gratefully acknowledged.  The GSM
 description was adopted from the IMTC Voice over IP Forum Service
 Interoperability Implementation Agreement (January 1997).  Fred Burg
 and Terry Lyons helped with the G.729 description.

Schulzrinne & Casner Standards Track [Page 42] RFC 3551 RTP A/V Profile July 2003

15. Intellectual Property Rights Statement

 The IETF takes no position regarding the validity or scope of any
 intellectual property or other rights that might be claimed to
 pertain to the implementation or use of the technology described in
 this document or the extent to which any license under such rights
 might or might not be available; neither does it represent that it
 has made any effort to identify any such rights.  Information on the
 IETF's procedures with respect to rights in standards-track and
 standards-related documentation can be found in BCP-11.  Copies of
 claims of rights made available for publication and any assurances of
 licenses to be made available, or the result of an attempt made to
 obtain a general license or permission for the use of such
 proprietary rights by implementors or users of this specification can
 be obtained from the IETF Secretariat.
 The IETF invites any interested party to bring to its attention any
 copyrights, patents or patent applications, or other proprietary
 rights which may cover technology that may be required to practice
 this standard.  Please address the information to the IETF Executive
 Director.

16. Authors' Addresses

 Henning Schulzrinne
 Department of Computer Science
 Columbia University
 1214 Amsterdam Avenue
 New York, NY 10027
 United States
 EMail: schulzrinne@cs.columbia.edu
 Stephen L. Casner
 Packet Design
 3400 Hillview Avenue, Building 3
 Palo Alto, CA 94304
 United States
 EMail: casner@acm.org

Schulzrinne & Casner Standards Track [Page 43] RFC 3551 RTP A/V Profile July 2003

17. Full Copyright Statement

 Copyright (C) The Internet Society (2003).  All Rights Reserved.
 This document and translations of it may be copied and furnished to
 others, and derivative works that comment on or otherwise explain it
 or assist in its implementation may be prepared, copied, published
 and distributed, in whole or in part, without restriction of any
 kind, provided that the above copyright notice and this paragraph are
 included on all such copies and derivative works.  However, this
 document itself may not be modified in any way, such as by removing
 the copyright notice or references to the Internet Society or other
 Internet organizations, except as needed for the purpose of
 developing Internet standards in which case the procedures for
 copyrights defined in the Internet Standards process must be
 followed, or as required to translate it into languages other than
 English.
 The limited permissions granted above are perpetual and will not be
 revoked by the Internet Society or its successors or assigns.
 This document and the information contained herein is provided on an
 "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
 TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
 BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
 HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
 MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.

Acknowledgement

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

Schulzrinne & Casner Standards Track [Page 44]

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