Internet Engineering Task Force                                   AVT WG
Internet Draft                                               Schulzrinne
ietf-avt-profile-new-00.txt                                      Columbia U.
March 26, 1997
Expires: September 9, 1997

    RTP Profile for Audio and Video Conferences with Minimal Control


   This document is an Internet-Draft. Internet-Drafts are working
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         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, 13, 16, 17, 18 and
   34. The PostScript version of this draft contains change bars.

   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 DS 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

   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

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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

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

   String-to-key mapping: A user-provided string ("pass phrase") is

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        hashed with the MD5 algorithm to a 16-octet digest. An

        -bit key is extracted from the digest by taking the first

        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. (This does not preclude the use of
        these definitions when RTP is carried by other lower-layer

   Transport mapping: The standard mapping of RTP and RTCP to

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        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:

        o name 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;

        o indication of who has change control over the encoding (for
         example, ISO, CCITT/ITU, other international standardization
         bodies, a consortium or a particular company or group of

        o any operating parameters or profiles;

        o a 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;

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

        o the 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 drawn from the

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

   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

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   assign proprietary encodings to particular, fixed payload types in
   the range reserved for dynamic payload types.

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

        o The encoding is of interest to the Internet community at

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

        o The description is sufficient to build a decoder.

   The four-character encoding names are those those by the Session
   Description Protocol (SDP) (RFC XXXX) .

4 Audio

4.1 Encoding-Independent Rules

   For applications which send no 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. 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

   -channel encodings, each sampling period (say,

   of a second) generates

   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.

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   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. (The Apple Macintosh
   computers have native sample rates of 22254.54 and 11127.27, which
   can be converted to 22050 and 11025 with acceptable quality by
   dropping 4 or 2 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
   have the same length measured in octets. This does not work when
   carrying frames of different sizes unless the frame sizes are

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   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

     encoding    sample/frame    bits/sample    ms/frame    ms/packet
     1016        frame           N/A                  30           30
     DVI4        sample          4                                 20
     G721        sample          4                                 20
     G722        sample          8                                 20
     G723        frame           N/A                  30           30
     G728        frame           N/A                 2.5           20
     G729        frame           N/A                  10           20
     GSM         frame           N/A                  20           20
     L8          sample          8                                 20
     L16         sample          16                                20
     LPC         frame           N/A                  20           20
     MPA         frame           N/A                               20
     PCMA        sample          8                                 20
     PCMU        sample          8                                 20
     VDVI        sample          var.                              20

   Table 1: Properties of Audio Encodings

   The characteristics of standard audio encodings are shown in Table 1
   and their payload types are listed in Table 4.

4.5.1 1016

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

   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.

4.5.2 CN

   The G.764-based VAD (voice activity detector) noise level packet
   contains a single-octet message to the receiver to play comfort noise
   at the absolute dBmO level specified by the G.764 level index. 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
   mapping of the index to absolute noise levels measured on the
   transmit side is given in Table 2, with the level index packed into
   the least significant bits of the noise-level payload, as shown

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

   The RTP header for the comfort noise packet should be constructed as
   if the VAD noise were an independent codec, but sharing the media
   clock and sequence number space with the associated voice codec.
   Thus, the RTP timestamp designates the beginning of the silence
   period, using the timestamp frequency of the payload type immediately
   preceding the CN packet. The RTP packet should not have the marker
   bit set.

   Note: dBrnc0 is the noise power measured in dBrnC, but referenced to
   the zero-level transmission level point (TLP). Typically, the two-
   wire interface in telephony is at the zero-level TLP of 0 dBm. dBrnC
   is the power level of noise with C-message weighting expressed in
   decibels relative to reference noise. Reference noise power is -90

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                       Index    Noise Level (dBrncO)
                           0               Idle Code
                           1                    16.6
                           2                    19.7
                           3                    22.6
                           4                    24.9
                           5                    26.9
                           6                    29.0
                           7                    31.0
                           8                    32.8
                           9                    34.6
                          10                    36.2
                          11                    37.9
                          12                    39.7
                          13                    41.6
                          14                    43.8
                          15                    46.6

   Table 2: G.764 noise level mapping

   dBm or 1 pW.  (dBm is the power level in decibels relative to 1 mW,
   with an impedance of 600 Ohms.) The C-message weighting is described
   in [6]. To obtain dBmC0 levels, subtract 90 dB from the values

4.5.3 DVI4

   DVI4 is specified, with pseudo-code, in [7] as the IMA ADPCM wave

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

        o The header contains the predicted value rather than the first
         sample value.

        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

        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

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         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.

   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 G721

   G721 is specified in ITU recommendation G.721. Reference
   implementations for G.721 are available as part of the CCITT/ITU-T
   Software Tool Library (STL) from the ITU General Secretariat, Sales
   Service, Place du Nations, CH-1211 Geneve 20, Switzerland. The
   library is covered by a license.

4.5.5 G722

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

4.5.6 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". Audio is encoded in 30 ms frames, with an additional delay

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   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

   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.

4.5.7 G726-32

   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 is a backwards-compatible superset of G.721, a recommendation
   which is no longer in force. G.726 also describes codecs operating at
   40 (5 bits/sample), 24 (3 bits/sample) and 16 kb/s (2 bits/sample).
   These are labeled G726-40, G726-24 and G726-16, respectively.

   No header information shall be included as part of the audio data.
   The 4-bit code words of the G.726 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
   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.

4.5.8 G728

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   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-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.

   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.

             1         2         3        3
   0         0         0         0        9
   <--------------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.

4.5.9 G729

   G.729 and G.729A are defined in ITU-T Recommendation G.729, "Coding
   of Speech at 8 kbit/s using Conjugate Structure-Algebraic Code
   Excited Linear Predictive (CS-ACELP) Coding" and its Annex A,
   respectively.  These two audio codecs are compatible with each other
   on the wire so there is no need to distinguish further between them.
   The codecs were optimized to represent speech with a high quality;
   G.729A achieves this with very low complexity.

   A voice activity detector (VAD) and comfort noise generator (CNG) is
   defined in G.729 Annex B (G.729B). It can be used in conjunction with
   either G.729 or G.729A. A G.729 or G.729A frame contains 10 octets,
   while the G.729B comfort noise frame contains 4 octets.

   An RTP packet may consist of zero or more G.729 or G.729A frames,
   followed by zero or one G.729B payload.

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   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|
   | |             |         |         |               | |         |

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

   |   C2    |  S2   | GA2 |  GB2  |
   |         |       |     |       |
   |8 9 1 1 1|0 1 2 3|0 1 2|0 1 2 3|
   |    0 1 2|       |     |       |

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 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

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   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 (4-7) and the 4 most significant bits of F1 (2-5) in the 4 least
   significant bits (0-3). The second octet contains the 2 least bits of
   F1 in bits 6-7, and F2 in bits 0-5, 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.

4.5.12 L16

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


   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

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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 3: Ordering of GSM variables

   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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   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

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...").

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4.5.15 PCMA

   PCMA is specified in CCITT/ITU-T recommendation G.711. Audio data is
   encoded as eight bits per sample, after logarithmic scaling. Code to
   convert between linear and A-law companded data is available in [7].
   A detailed description is given by Jayant and Noll [12].

4.5.16 PCMU

   PCMU is specified in CCITT/ITU-T recommendation G.711. Audio data is
   encoded as eight bits per sample, after logarithmic scaling. Code to
   convert between linear and mu-law companded data is available in [7].
   PCMU is the encoding used for the Internet media type audio/basic. A
   detailed description is given by Jayant and Noll [12].

4.5.17 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.

4.5.18 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.

   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

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                                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].

5.3 H261

   The encoding is specified in CCITT/ITU-T standard H.261. The
   packetization and RTP-specific properties are described in RFC 2032

5.4 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.

5.5 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

5.6 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:

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

6 Payload Type Definitions

   Table 4 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

   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. 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 4 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.

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   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 4.

7 Port Assignment

   As specified in the RTP protocol definition, RTP data is to be
   carried on an even UDP 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 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

8 Bibliography

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

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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          G721          A             8000          1
      3          GSM           A             8000          1
      4          G.723.1       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         G723          A             8000          1
      13         CN            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--22     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 4: Payload types (PT) for standard audio and video encodings

   [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] J. Bellamy, Digital Telephony New York: John Wiley & Sons, 1991.

   [7] 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.

   [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
   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

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   Engineering Task Force, Oct. 1996.

9 Acknowledgements

   The comments and careful review of Steve Casner are gratefully
   acknowledged. The GSM description was adopted from the IMTC Voice
   over IP Forum Service Interoperability Implementation Agreement
   (January 1997). Fred Burg helped with the G.729 description.

10 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


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



   An implementation is available at



   An implementation is available from Jack Jansen at



   An implementation is available at

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   Reference implementations for G.723.1 are available as part of the
   CCITT/ITU-T Software Tool Library (STL) from the ITU General
   Secretariat, Sales Service, Place du Nations, CH-1211 Geneve 20,
   Switzerland. The library is covered by a license.

   The specification also contains C source code. Source code files are
   available at

   and test vectors at


   Reference implementations for G.729, G.729A and G.729B are available
   as part of the ITU-T Software Tool Library from the ITU General
   Secretariat, Sales Service, Place de Nations, CH-1211 Geneve 20,
   Switzerland. The library is covered by a license.


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



   An implementation is available at


Schulzrinne                                                  [Page 26]