Internet Engineering Task Force                                   AVT WG
Internet Draft                                               Schulzrinne
ietf-avt-profile-new-02.txt                                  Columbia U.
November 20, 1997
Expires: January 1, 1998

    RTP Profile for Audio and Video Conferences with Minimal Control


   This document is an Internet-Draft. Internet-Drafts are working
   documents of the Internet Engineering Task Force (IETF), its areas,
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   Distribution of this document is unlimited.


         This memo 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

         The 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. However,

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         the encoding definitions are independent of the
         particular transport mechanism used. 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 draft revises RFC 1890. It is fully backwards-compatible with
   RFC 1890 and codifies existing practice. It is intended that this
   draft form the basis of a new RFC to obsolete RFC 1890 as it moves to
   Draft Standard.

   Besides wording clarifications and filling in RFC numbers for payload
   type definitions, this draft adds payload types 4, 16, 17, 18, 19 and
   34. The PostScript version of this draft contains change bars marking
   changes make since draft -00.

   A tentative TCP encapsulation is defined.

   According to Peter Hoddie of Apple, only pre-1994 Macintosh used the
   22254.54 rate and none the 11127.27 rate.

   Note to RFC editor: This section is to be removed before publication
   as an RFC. All RFC TBD should be filled in with the number of the RTP
   specification RFC submitted for Draft Standard status.

1 Introduction

   This profile defines aspects of RTP left unspecified in the RTP
   Version 2 protocol definition (RFC XXXX). 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 occurs by use of the appropriate applications;
   there is no explicit indication by port number, protocol identifier
   or the like. Applications such as session directories should refer to
   this profile as "RTP/AVP".

   Other profiles may make different choices for the items specified

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   This document also defines a set of payload formats for audio.

   This draft defines the term media type as dividing encodings of audio
   and video content into three classes: audio, video and audio/video

2 RTP and RTCP Packet Forms and Protocol Behavior

   The section "RTP Profiles and Payload Format Specification" of RFC
   TBD 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

   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

   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.

   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 is sent every
        reporting interval, other items should be sent only 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

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   Security: The RTP default security services are also the default
        under this profile.

   String-to-key mapping: A user-provided string ("pass phrase") is
        hashed with the MD5 algorithm to a 16-octet digest. An !n!-bit
        key is extracted from the digest by taking the first !n! bits
        from the digest. If several keys are needed with a total length
        of 128 bits or less (as for triple DES), they are extracted in
        order from that digest. The octet ordering is specified in RFC
        1423, Section 2.2. (Note that some DES implementations require
        that the 56-bit key be expanded into 8 octets by inserting an
        odd parity bit in the most significant bit of the octet to go
        with each 7 bits of the key.)

   It is suggested that pass phrases are restricted to ASCII letters,
   digits, the hyphen, and white space to reduce the the chance of
   transcription errors when conveying keys by phone, fax, telex or

   The pass phrase may be preceded by a specification of the encryption
   algorithm. Any characters up to the first slash (ASCII 0x2f) are
   taken as the name of the encryption algorithm. The encryption format
   specifiers should be drawn from RFC 1423 or any additional
   identifiers registered with IANA. If no slash is present, DES-CBC is
   assumed as default. The encryption algorithm specifier is case

   The pass phrase typed by the user is transformed to a canonical form
   before applying the hash algorithm. For that purpose, we define
   return, tab, or vertical tab as well as all characters contained in
   the Unicode space characters table. The transformation consists of
   the following steps: (1) convert the input string to the ISO 10646
   character set, using the UTF-8 encoding as specified in Annex P to
   ISO/IEC 10646-1:1993 (ASCII characters require no mapping, but ISO
   8859-1 characters do); (2) remove leading and trailing white space
   characters; (3) replace one or more contiguous white space characters
   by a single space (ASCII or UTF-8 0x20); (4) convert all letters to
   lower case and replace sequences of characters and non-spacing
   accents with a single character, where possible. A minimum length of
   16 key characters (after applying the transformation) should be
   enforced by the application, while applications must allow up to 256
   characters of input.

   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

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        other lower-layer protocols.)

   Transport mapping: The standard mapping of RTP and RTCP to
        transport-level addresses is used.

   Encapsulation: No encapsulation of RTP packets is specified.

3 Registering Payload Types

   This profile defines a set of standard encodings and their payload
   types when used within RTP. Other encodings and their payload types
   are to be registered with the Internet Assigned Numbers Authority
   (IANA). When registering a new encoding/payload type, the following
   information should be provided:

        oname and description of encoding, in particular the RTP
         timestamp clock rate; the names defined here are 3 or 4
         characters long to allow a compact representation if needed;

        oindication of who has change control over the encoding (for
         example, ISO, ITU-T, other international standardization
         bodies, a consortium or a particular company or group of

        oany operating parameters or profiles;

        oa reference to a further description, if available, for example
         (in order of preference) an RFC, a published paper, a patent
         filing, a technical report, documented source code or a
         computer manual;

        ofor proprietary encodings, contact information (postal and
         email address);

        othe payload type value for this profile, if necessary (see

   Note that not all encodings to be used by RTP need to be assigned a
   static payload type. Non-RTP means beyond the scope of this memo
   (such as directory services or invitation protocols) may be used to
   establish a dynamic mapping between a payload type and an encoding
   ("dynamic payload types"). Applications should first use the range 96
   to 127 for dynamic payload types. Only applications which need to
   define more than 32 dynamic payload types may redefine codes below
   96. 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.

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   Note that 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
   assign proprietary encodings to particular, fixed payload types in
   the range reserved for dynamic payload types. SDP (RFC XXXX ) defines
   such a mapping mechanism.

   The available payload type space is relatively small. Thus, new
   static payload types are assigned only if the following conditions
   are met:

        oThe encoding is of interest to the Internet community at large.

        oIt offers benefits compared to existing encodings and/or is
         required for interoperation with existing, widely deployed
         conferencing or multimedia systems.

        oThe description is sufficient to build a decoder.

   For implementor convenience, this profile contains descriptions of
   encodings which do not currently have a static payload type assigned
   to them.

   The Session Description Protocol (SDP) (RFC XXXX)  uses the encoding
   names defined here.

4 Audio

4.1 Encoding-Independent Rules

   For applications which send either no packets or comfort-noise
   packets during silence, the first packet of a talkspurt, that is, the
   first packet after a silence period, is distinguished by setting the
   marker bit in the RTP data header to one. The marker bits 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 set the bit to zero.

   The RTP clock rate used for generating the RTP timestamp is
   independent of the number of channels and the encoding; it equals the
   number of sampling periods per second. For !N!-channel encodings,
   each sampling period (say, 1/8000 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

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   lower-numbered channels precedes that from higher-numbered channels.
   For more than two channels, the convention followed by the AIFF-C
   audio interchange format should be followed [1], using the following

   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           quadrophonic      Fl       Fr    Rl    Rr
   4                              l       c     r     S
   5                             Fl       Fr    Fc    Sl    Sr
   6                              l       lc    c     r     rc    S

   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: 8000, 11025,
   16000, 22050, 24000, 32000, 44100 and 48000 Hz. (Older Apple
   Macintosh computers had a native sample rate of 22254.54 Hz, which
   can be converted to 22050 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
   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

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   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
   links with severe bandwidth constraints, a higher packetization delay
   may be appropriate. A receiver should accept packets representing
   between 0 and 200 ms of audio data. (For framed audio encodings, a
   receiver should accept packets with 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
   are 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 are
   transmitted in network byte order (i.e., most significant octet

   The packing of sample-based encodings producing less than one octet
   per sample is encoding-specific.

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 since the audio frame duration (in
   octets) is defined as part of the encoding, as long as all frames

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   have the same length measured in octets. This does not work when
   carrying frames of different sizes unless the frame sizes are
   relatively prime.

   For frame-based codecs, the channel order is defined for the whole
   block. That is, for two-channel audio, right and left samples are
   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 capturing time of the first sample in
   the first frame, that is, the oldest information in the packet.

4.5 Audio Encodings

   The characteristics of standard audio encodings are shown in Table 1;
   those assigned static payload types are listed in Table 3. 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. Non-RTP means MUST
   indicate the appropriate sampling rate.

4.5.1 1016

   Encoding 1016 is a frame based encoding using code-excited linear
   prediction (CELP) and is specified in Federal Standard FED-STD 1016

4.5.2 CN

   The CN (comfort noise) packet contains a single-octet message to the
   receiver to play comfort noise at the absolute level specified. This
   message would normally be sent once at the beginning of a silence
   period (which also indicates the transition from speech to silence),
   but rate of noise level updates is implementation specific. The
   magnitude of the noise level is packed into the least significant
   bits of the noise-level payload, as shown below.

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   name of                                    sampling                  default
   encoding    sample/frame    bits/sample        rate    ms/frame    ms/packet
   1016        frame           N/A               8,000          30           30
   CN          frame           N/A                var.
   DVI4        sample          4                  var.                       20
   G722        sample          8                16,000                       20
   G723        frame           N/A               8,000          30           30
   G726-16     sample          2                 8,000                       20
   G726-24     sample          3                 8,000                       20
   G726-32     sample          4                 8,000                       20
   G726-40     sample          5                 8,000                       20
   G727-16     sample          2                 8,000                       20
   G727-24     sample          3                 8,000                       20
   G727-32     sample          4                 8,000                       20
   G727-40     sample          5                 8,000                       20
   G728        frame           N/A               8,000         2.5           20
   G729        frame           N/A               8,000          10           20
   GSM         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.          20
   PCMA        sample          8                  var.          20
   PCMU        sample          8                  var.          20
   SX7300P     frame           N/A               8,000          15           30
   SX8300P     frame           N/A               8,000          15           30
   VDVI        sample          var.               var.          20

   Table 1: Properties of Audio Encodings (N/A:  not  applicable;  var.:

   The noise level is expressed in dBov, with values from 0 to 127 dBov.
   dBov is the level relative to the overload of the system. (Note:
   Representation relative to the overload point of a system is
   particularly useful for digital implementations, since one does not
   need to know the relative calibration of the analog circuitry.)
   Example: In 16-bit linear PCM system (L16), a signal with 0 dBov
   represents a square wave with the maximum possible amplitude (+/-
   32767). -63 dBov corresponds to -58 dBm0 in a standard telephone
   system. (dBm is the power level in decibels relative to 1 mW, with an
   impedance of 600 Ohms.)

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      0 1 2 3 4 5 6 7
     |0|   level     |

   The RTP header for the comfort noise packet should be constructed as
   if the comfort noise were an independent codec. Thus, the RTP
   timestamp designates the beginning of the silence period. A static
   payload type is assigned for a sampling rate of 8,000 Hz; if other
   sampling rates are needed, they should be defined through dynamic
   payload types. The RTP packet should not have the marker bit set.

   The CN payload type is primarily for use with L16, DVI4, PCMA, PCMU
   and other audio codecs that do not support comfort noise as part of
   the codec itself. G.723.1 and G.729 have their own comfort noise
   systems as part of Annexes A (G.723.1) and B (G.729), respectively.

4.5.3 DVI4

   DVI4 is specified, with pseudo-code, in [6] as the IMA ADPCM wave

   However, the encoding defined here as DVI4 differs in three respects
   from this recommendation:

        oThe RTP DVI4 header contains the predicted value rather than
         the first sample value contained the IMA ADPCM block header.

        oIMA 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

        oFor 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 little-endian order.

   Each packet contains a single DVI block. This profile only defines
   the 4-bit-per-sample version, while IMA also specifies a 3-bit-per-
   sample encoding.

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   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.

   Packing of samples for multiple channels is for further study.

   The document IMA Recommended Practices for Enhancing Digital Audio
   Compatibility in Multimedia Systems (version 3.0) contains the
   algorithm description. It is available from

   Interactive Multimedia Association
   48 Maryland Avenue, Suite 202
   Annapolis, MD 21401-8011
   phone: +1 410 626-1380

4.5.4 G722

   G722 is specified in ITU-T Recommendation G.722, "7 kHz audio-coding
   within 64 kbit/s".

4.5.5 G723

   G.723.1 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 an encoded signal bit-error sensitivity specification 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

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   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

   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. This coder was optimized to represent speech
   with near-toll quality at the above rates using a limited amount of

   All the bits of the encoded bit stream are transmitted always from
   the the least significant bit towards the most significant bit.

4.5.6 G726-16, G726-24, G726-32, G726-40

   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 8000 samples/sec to and from a 32 kbit/s channel.
   The conversion is applied to the PCM stream using an Adaptive
   Differential Pulse Code Modulation (ADPCM) transcoding technique.
   G.726 describes codecs operating at 16 kb/s (2 bits/sample), 24 kb/s
   (3 bits/sample), 32 kb/s (4 bits/sample), 40 kb/s (5 bits/sample).
   These encodings are labeled G726-16, G726-24, G726-32 and G726-40,

   Note: In 1990, ITU-T Recommendation G.721 was merged with
   Recommendation G.723 into ITU-T Recommendation G.726. Thus, G726-32
   designates the same algorithm as G721 in RFC 1890.

   No header information shall be included as part of the audio data.
   The 4-bit code words of the G726-32 encoding MUST be packed into
   octets as follows: the first code word is placed in the four least
   significant bits of the first octet, with the least significant bit
   of the code word in the least significant bit of the octet; the
   second code word is placed in the four most significant bits of the
   first octet, with the most significant bit of the code word in the
   most significant bit of the octet. Subsequent pairs of the code words

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   shall be packed in the same way into successive octets, with the
   first code word of each pair placed in the least significant four
   bits of the octet. It is prefered that the voice sample be extended
   with silence such that the encoded value comprises an even number of
   code words. [TBD: Shouldn't we just require an even number of

4.5.7 G727-16, G727-24, G727-32, G727-40

   ITU-T Recommendation G.727, "5-, 4-, 3- and 2-bits sample embedded
   adaptive differential pulse code modulation (ADPCM)", specifies an
   embedded ADPCM algorithm which has the intrinsic capability of
   dropping bits in the encoded words to alleviate network congestion
   conditions.  The algorithm, although not bitstream compatible with
   G.726, was based and has a structure similar to the G.726 ADPCM

4.5.8 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. For
   example:  octet B1 contains the eight most significant bits of vector
   V1, the MSB of V1 is MSB of B1.

             1         2         3        3
   0         0         0         0        9
   <---V1---><---V2---><---V3---><---V4---> vectors
   <--B1--><--B2--><--B3--><--B4--><--B5--> octets
   <------------- frame 1 ---------------->

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   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.

4.5.9 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 complexity-reduced 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. 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 [7].

   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:

    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|0 1 2 3 4|0 1 2 3|0 1 2 3 4|S|
   |0|         |       |         |V|    RESV = Reserved (zero)

   An 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 payloads. The presence
   of a comfort noise frame can be deduced from the length of the RTP

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   A floating-point version of the G.729, G.729 Annex A, and G.729 Annex
   B will be available shortly as Annex C to Recommendation G.729.

   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. Bits are numbered
   as Internet order, that is, the most significant bit is bit 0.

    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|
   | |             |         |         |               | |         |

                   4                   5                   6
   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 2 3
   |       C1      |  S1   | GA1 |  GB1  |    P2   |      C2       |
   |               |       |     |       |         |               |
   |5 6 7 8 9 1 1 1|0 1 2 3|0 1 2|0 1 2 3|0 1 2 3 4|0 1 2 3 4 5 6 7|
   |          0 1 2|       |     |       |         |               |

   4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9
   |   C2    |  S2   | GA2 |  GB2  |
   |         |       |     |       |
   |8 9 1 1 1|0 1 2 3|0 1 2|0 1 2 3|
   |    0 1 2|       |     |       |

   The encoding name "G729B" is assigned for the case when a particular
   RTP payload type is to contain G.729 Annex B comfort noise packets
   only.  This may be necessary if the underlying RTP mechanism has no
   means of distinguishing talkspurt from comfort-noise packets.

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4.5.10 GSM

   GSM (group speciale mobile) denotes the European GSM 06.10
   provisional standard for full-rate speech transcoding, prI-ETS 300
   036, which is based on RPE/LTP (residual pulse excitation/long term
   prediction) coding at a rate of 13 kb/s [8,9,10]. The text of the
   standard can be obtained from

   ETSI (European Telecommunications Standards Institute)
   ETSI Secretariat: B.P.152
   F-06561 Valbonne Cedex
   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. General Packaging Issues

   The GSM standard specifies the bit stream produced by the codec, but
   does not specify how these bits should be packed for transmission.
   Some software implementations of the GSM codec use a different
   packing than that specified here.

   In the GSM encoding used by RTP, the bits are 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 as follows: GSM variable names and numbers

   So if F.i signifies the ith bit of the field F, and bit 0 is the most
   significant bit, and the bits of every octet are numbered from 0 to 7
   from most to least significant, then in the RTP encoding we have:

4.5.11 L8

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

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        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

4.5.12 L16

   L16 denotes uncompressed audio data, using 16-bit signed
   representation with 65535 equally divided steps between minimum and

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   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

   maximum signal level, ranging from --32768 to 32767. The value is
   represented in two's complement notation and network byte order.

4.5.13 LPC

   LPC designates an experimental linear predictive encoding contributed
   by Ron Frederick, Xerox PARC, which is based on an implementation
   written by Ron Zuckerman, Motorola, 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.

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4.5.14 MPA

   MPA denotes MPEG-I or MPEG-II 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 2038 [11].

   Sampling rate and channel count are contained in the payload. MPEG-I
   audio supports sampling rates of 32000, 44100, and 48000 Hz (ISO/IEC
   11172-3, section 1.1; "Scope"). MPEG-II additionally supports ISO/IEC
   11172-3 Audio. "TBD"). [Something missing here.]

4.5.15 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 [12]. 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 signficant 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 G.711 shall always be
   transmitted as 8-bit samples.

4.5.16 RED

   The redundant audio payload format "RED" is specified by RFC XXX. 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 the
   previous packetization interval. This allows an approximation of the
   data from lost packets to be recovered upon decoding of the following
   packet, giving much improved sound quality when compared with silence
   substitution for lost packets.

4.5.17 SX7300P

   The SX7300P is a low-complexity CELP-based audio codec operating at a
   sampling rate of 8000 Hz. It encodes blocks of 120 audio samples (15
   ms) into an encoded frame of 14 octets, yielding an encoded bit rate
   of approximately 7467 b/s.

4.5.18 SX8300P

   The SX8300P is a low-complexity CELP-based audio codec operating at a
   sampling rate of 8000 Hz. It encodes blocks of 120 audio samples (15

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   ms) into an encoded frame of 16 octets, yielding an encoded bit rate
   of approximately 8533 b/s.

4.5.19 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

   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

5 Video

   The following video encodings are currently defined, with their
   abbreviated names used for identification:

5.1 CelB

   The CELL-B encoding is a proprietary encoding proposed by Sun
   Microsystems. The byte stream format is described in RFC 2029 [13].

5.2 JPEG

   The encoding is specified in ISO Standards 10918-1 and 10918-2. The
   RTP payload format is as specified in RFC 2035 [14].

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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 [15].

5.4 H263

   The encoding is specified in ITU-T Recommendation H.263, "Video
   coding for low bit rate communication". The packetization and RTP-
   specific properties are described in [16].

5.5 MPV

   MPV designates the use MPEG-I and MPEG-II 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 2038
   [11], Section 3.

5.6 MP2T

   MP2T designates the use of MPEG-II transport streams, for either
   audio or video. The encapsulation is described in RFC 2038 [11],
   Section 2. See the description of the MPA audio encoding for contact

5.7 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
   Xerox Palo Alto Research Center
   3333 Coyote Hill Road
   Palo Alto, CA 94304
   United States
   electronic mail:

6 Payload Type Definitions

   Table 3 defines this profile's static payload type values for the PT
   field of the RTP data header. A new RTP payload format specification
   may be registered with the IANA by name, and may also be assigned a
   static payload type value from the range marked in Section 3.

   In addition, payload type values in the range 96--127 may be defined
   dynamically through a conference control protocol, which is beyond

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   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. The payload type range
   marked 'reserved' has been set aside so that RTCP and RTP packets can
   be reliably distinguished (see Section "Summary of Protocol
   Constants" of the RTP protocol specification).

   An RTP source emits a single RTP payload type at any given instant.
   The interleaving or multiplexing of several RTP media types within a
   single RTP session is not allowed, but multiple RTP sessions may be
   used in parallel to send multiple media types. An RTP source may
   change payload types during a session.

   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. A single RTP session
   consists of payload types of one and only media type.

   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. The media
   types in Table 3 are marked as "A" for audio, "V" for video and "AV"
   for combined audio/video streams.

   Audio applications operating under this profile should, at minimum,
   be able to send and receive payload types 0 (PCMU) and 5 (DVI4). This
   allows interoperability without format negotiation and successful
   negotation with a conference control protocol.

   All current video encodings use a 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
   are possible.  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.

   The standard video encodings and their payload types are listed in
   Table 3.

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      PT         encoding      media type    clock rate    channels
                 name                        (Hz)          (audio)
      0          PCMU          A             8000          1
      1          1016          A             8000          1
      2          G726-32       A             8000          1
      3          GSM           A             8000          1
      4          G723          A             8000          1
      5          DVI4          A             8000          1
      6          DVI4          A             16000         1
      7          LPC           A             8000          1
      8          PCMA          A             8000          1
      9          G722          A             16000         1
      10         L16           A             44100         2
      11         L16           A             44100         1
      12         unassigned    A
      13         unassigned    A
      14         MPA           A             90000         (see text)
      15         G728          A             8000          1
      16         DVI4          A             11025         1
      17         DVI4          A             22050         1
      18         G729          A             8000          1
      19         CN            A             8000          1
      20         unassigned    A
      21         unassigned    A
      22         unassigned    A
      23         unassigned    A
      24         unassigned    V
      25         CelB          V             90000
      26         JPEG          V             90000
      27         unassigned    V
      28         nv            V             90000
      29         unassigned    V
      30         unassigned    V
      31         H261          V             90000
      32         MPV           V             90000
      33         MP2T          AV            90000
      34         H263          V             90000
      35--71     unassigned    ?
      72--76     reserved      N/A           N/A           N/A
      77         RED           A             N/A           N/A
      78--95     unassigned    ?
      96--127    dynamic       ?

   Table 3: Payload types (PT) for standard audio and video encodings

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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. If the application does not define its
   own method of delineating RTP and RTCP packets, it SHOULD prefix each
   packet with a two-octet length field.

   (Note: RTSP [17] provides its own encapsulation and does not need an
   extra length indication.)

8 Port Assignment

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

   Applications operating under this profile may use any such UDP or TCP
   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 accomodate port number allocation
   practice within 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

9 Bibliography

   [1] Apple Computer, "Audio interchange file format AIFF-C," Aug.
   1991.  (also

   [2] Office of Technology and Standards, "Telecommunications: Analog
   to digital conversion of radio voice by 4,800 bit/second code excited
   linear prediction (celp)," Federal Standard FS-1016, GSA, Room 6654;
   7th & D Street SW; Washington, DC 20407 (+1-202-708-9205), 1990.

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   [3] J. P. Campbell, Jr., T. E. Tremain, and V. C. Welch, "The
   proposed Federal Standard 1016 4800 bps voice coder: CELP," Speech
   Technology , vol. 5, pp. 58--64, April/May 1990.

   [4] J. P. Campbell, Jr., T. E. Tremain, and V. C. Welch, "The federal
   standard 1016 4800 bps CELP voice coder," Digital Signal Processing ,
   vol. 1, no. 3, pp. 145--155, 1991.

   [5] J. P. Campbell, Jr., T. E. Tremain, and V. C. Welch, "The dod 4.8
   kbps standard (proposed federal standard 1016)," in Advances in
   Speech Coding (B. Atal, V. Cuperman, and A. Gersho, eds.), ch. 12,
   pp. 121--133, Kluwer Academic Publishers, 1991.

   [6] IMA Digital Audio Focus and Technical Working Groups,
   "Recommended practices for enhancing digital audio compatibility in
   multimedia systems (version 3.00)," tech. rep., Interactive
   Multimedia Association, Annapolis, Maryland, Oct. 1992.

   [7] D. Deléam 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, Oct. 1996.

   [8] M. Mouly and M.-B. Pautet, The GSM system for mobile
   communications Lassay-les-Chateaux, France: Europe Media Duplication,

   [9] J. Degener, "Digital speech compression," Dr. Dobb's Journal ,
   Dec.  1994.

   [10] S. M. Redl, M. K. Weber, and M. W. Oliphant, An Introduction to
   GSM Boston: Artech House, 1995.

   [11] D. Hoffman, G. Fernando, and V. Goyal, "RTP payload format for
   MPEG1/MPEG2 video," Request for Comments (Proposed Standard) RFC
   2038, Internet Engineering Task Force, Oct. 1996.

   [12] N. S. Jayant and P. Noll, Digital Coding of Waveforms--
   Principles and Applications to Speech and Video Englewood Cliffs, New
   Jersey: Prentice-Hall, 1984.

   [13] M. Speer and D. Hoffman, "RTP payload format of sun's CellB
   video encoding," Request for Comments (Proposed Standard) RFC 2029,
   Internet Engineering Task Force, Oct. 1996.

   [14] L. Berc, W. Fenner, R. Frederick, and S. McCanne, "RTP payload
   format for JPEG-compressed video," Request for Comments (Proposed

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   Standard) RFC 2035, Internet Engineering Task Force, Oct. 1996.

   [15] T. Turletti and C. Huitema, "RTP payload format for H.261 video
   streams," Request for Comments (Proposed Standard) RFC 2032, Internet
   Engineering Task Force, Oct. 1996.

   [16] C. C. Zhu, "RTP payload format for H.263 video streams,"
   Internet Draft, Internet Engineering Task Force, Mar. 1997.  Work in

   [17] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming
   protocol (RTSP)," Internet Draft, Internet Engineering Task Force,
   July 1997.  Work in progress.

10 Acknowledgements

   The comments and careful review of Steve Casner, Simao Campos and
   Richard Cox 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.

11 Address of Author

   Henning Schulzrinne
   Dept. of Computer Science
   Columbia University
   1214 Amsterdam Avenue
   New York, NY 10027
   electronic mail:

   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 The
   ITU-T STL is covered by a license defined in ITU-T Recommendation
   G.191, " Software tools for speech and audio coding standardization


   Information on the UCS Transformation Format 8 (UTF-8) is available

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   The U.S. DoD's Federal-Standard-1016 based 4800 bps code excited
   linear prediction voice coder version 3.2 (CELP 3.2) Fortran and C
   simulation source codes are available for worldwide distribution at
   no charge (on DOS diskettes, but configured to compile on Sun SPARC
   stations) from:  Bob Fenichel, National Communications System,
   Washington, D.C. 20305, phone +1-703-692-2124, fax +1-703-746-4960.

   An implementation is also available at



   An implementation is available from Jack Jansen at



   An implementation of the G.722 algorithm is available as part of the
   ITU-T STL, described above.


   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.

   G726-16 through G726-40

   G726-16 through G726-40 are 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.

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   G727-16 through G727-40

   G727-16 through G727-40 are specified in the ITU-T Recommendation
   G.727, "5-, 4-, 3-, and 2-bit/sample embedded adaptive differential
   pulse code modulation". An implementation of the G.727 algorithm will
   be available in a future release of the ITU-T STL, described above.


   The reference C code implementation defining the G.729 algorithm and
   its Annexes A and B are available as an integral part of
   Recommendation G.729 from the ITU Sales Service, listed above. Both
   the algorithm and the C code are covered by a specific license. The
   contact information for obtaining the license is listed in the C


   A reference implementation was written by Carsten Borman and Jutta
   Degener (TU Berlin, Germany). It is available at


   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


   An implementation is available at



   An implementation of these algorithm is available as part of the
   ITU-T STL, described above. Code to convert between linear and mu-law
   companded data is also available in [6].

                           Table of Contents

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   1          Introduction ........................................    2
   2          RTP and RTCP Packet Forms and Protocol Behavior .....    3
   3          Registering Payload Types ...........................    5
   4          Audio ...............................................    6
   4.1        Encoding-Independent Rules ..........................    6
   4.2        Operating Recommendations ...........................    7
   4.3        Guidelines for Sample-Based Audio Encodings .........    8
   4.4        Guidelines for Frame-Based Audio Encodings ..........    8
   4.5        Audio Encodings .....................................    9
   4.5.1      1016 ................................................    9
   4.5.2      CN ..................................................    9
   4.5.3      DVI4 ................................................   11
   4.5.4      G722 ................................................   12
   4.5.5      G723 ................................................   12
   4.5.6      G726-16, G726-24, G726-32, G726-40 ..................   13
   4.5.7      G727-16, G727-24, G727-32, G727-40 ..................   14
   4.5.8      G728 ................................................   14
   4.5.9      G729 ................................................   15
   4.5.10     GSM .................................................   17   General Packaging Issues ............................   17   GSM variable names and numbers ......................   17
   4.5.11     L8 ..................................................   17
   4.5.12     L16 .................................................   18
   4.5.13     LPC .................................................   19
   4.5.14     MPA .................................................   20
   4.5.15     PCMA and PCMU .......................................   20
   4.5.16     RED .................................................   20
   4.5.17     SX7300P .............................................   20
   4.5.18     SX8300P .............................................   20
   4.5.19     VDVI ................................................   21
   5          Video ...............................................   21
   5.1        CelB ................................................   21
   5.2        JPEG ................................................   21
   5.3        H261 ................................................   22
   5.4        H263 ................................................   22
   5.5        MPV .................................................   22
   5.6        MP2T ................................................   22
   5.7        nv ..................................................   22
   6          Payload Type Definitions ............................   22
   7          RTP over TCP and Similar Byte Stream Protocols ......   25
   8          Port Assignment .....................................   25
   9          Bibliography ........................................   25
   10         Acknowledgements ....................................   27
   11         Address of Author ...................................   27

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