Internet Engineering Task Force                                   AVT WG
Internet Draft                                               Schulzrinne
ietf-avt-profile-new-04.txt                                  Columbia U.
November 18, 1998
Expires: May 18, 1999

    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,
   and its working groups.  Note that other groups may also distribute
   working documents as Internet-Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as ``work in progress''.

   To view the entire list of current Internet-Drafts, please check the
   ``1id-abstracts.txt'' listing contained in the Internet-Drafts Shadow
   Directories on (Africa), (Northern
   Europe), (Southern Europe), (Pacific
   Rim), (US East Coast), or (US West Coast).

   Distribution of this document is unlimited.


         This memorandum is a revision of RFC 1890 in preparation
         for advancement from Proposed Standard to Draft Standard
         status. Readers are encouraged to use the PostScript form
         of this draft to see where changes from RFC 1890 are
         marked by change bars. The revision process is not yet
         complete; some changes which have been discussed and
         tentatively accepted in meetings of the Audio/Video
         Transport working group have not yet been incorporated
         into this draft.

         This document describes a profile called 'RTP/AVP' for
         the use of the real-time transport protocol (RTP),
         version 2, and the associated control protocol, RTCP,
         within audio and video multiparticipant conferences with
         minimal control. It provides interpretations of generic
         fields within the RTP specification suitable for audio

Schulzrinne                                                   [Page 1]

Internet Draft                  Profile                November 18, 1998

         and video conferences. In particular, this document
         defines a set of default mappings from payload type
         numbers to encodings.

         This document also describes how audio and video data may
         be carried within RTP. It defines a set of standard
         encodings and their names when used within RTP. However,
         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 to the RFC.

   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 XXXX 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.

Schulzrinne                                                   [Page 2]

Internet Draft                  Profile                November 18, 1998

   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

2 RTP and RTCP Packet Forms and Protocol Behavior

   The section "RTP Profiles and Payload Format Specification" of RFC
   XXXX 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.

Schulzrinne                                                   [Page 3]

Internet Draft                  Profile                November 18, 1998

   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
        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

Schulzrinne                                                   [Page 4]

Internet Draft                  Profile                November 18, 1998

   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
        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 Additional Encodings with IANA

   This profile defines a set of encodings and assigns names to them. It
   is expected that additional encodings beyond this set will be defined
   in the future. These additional encodings may be registered with the
   Internet Assigned Numbers Authority (IANA) as explained here.

   It has been decided in discussions among the AVT and MMUSIC working
   groups and the Area Directors that the encoding names used in this
   profile should be registered as MIME subtype names under the "audio"
   and "video" MIME types. However, the procedures for doing this have
   not been established yet. This work must be completed before this
   draft will be ready for publication as an RFC.

   The MIME registration procedure needs to be extended to include
   additional information specifying how the encoding is used with RTP
   which is different from the information required to specify how an
   encoding is used in multimedia mail. Determining exactly what
   additional information is required is the open issue. At least the
   following information should be provided:

       o name of the encoding; the names defined here are 3 or 4
         characters long to allow a compact representation if needed;

       o a description of encoding, including in particular the RTP
         timestamp clock rate (or multiple rates for audio encodings
         with multiple sampling rates);

       o indication 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

Schulzrinne                                                   [Page 5]

Internet Draft                  Profile                November 18, 1998

       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);

   In addition to assigning names to encodings, this profile also also
   assigns static RTP payload types to some of them. However, the
   payload type number space is relatively small and cannot accommodate
   assignments for all existing and future encodings. During the early
   stages of RTP development, it was necessary to use statically
   assigned payload types because no other mechanism had been specified
   to bind encodings to payload types. It was anticipated that non-RTP
   means beyond the scope of this memo (such as directory services or
   invitation protocols) would be specified to establish a dynamic
   mapping between a payload type and an encoding. Now, mechanisms for
   defining dynamic payload type bindings have been specified in the
   Session Description Protocol (SDP), RFC 2327 [1], and in other
   protocols such as ITU-T recommendation H.323/H.245.  These mechanisms
   associate the registered name of the encoding/payload format, along
   with any additional required parameters such as the RTP timestamp
   clock rate and number of channels, to a payload type number.  This
   association is effective only for the duration of the RTP session in
   which the dynamic payload type binding is made. This association
   applies only to the RTP session for which it is made, thus the
   numbers can be re-used for different encodings in different sessions
   so the number space limitation is avoided.

   This profile reserves payload type numbers in the range 96-127
   exclusively for dynamic assignment. Applications should first use
   values in this range for dynamic payload types. Only applications
   which need to define more than 32 dynamic payload types may bind
   codes below 96, in which case it is RECOMMENDED that unassigned
   payload type numbers be used first. However, the statically assigned
   payload types are default bindings and may be dynamically bound to
   new encodings if needed. Redefining payload types below 96 may cause
   incorrect operation if an attempt is made to join a session without
   obtaining session description information that defines the dynamic
   payload types.

   Dynamic payload types should not be used without a well-defined
   mechanism to indicate the mapping. Systems that expect to
   interoperate with others operating under this profile should not make
   their own assignments of proprietary encodings to particular, fixed

Schulzrinne                                                   [Page 6]

Internet Draft                  Profile                November 18, 1998

   payload types.

   This specification establishes the policy that no additional static
   payload types will be assigned beyond the ones defined in this
   document. Establishing this policy avoids the problem of trying to
   create a set of criteria for accepting static assignments and
   encourages the implementation and deployment of the dynamic payload
   type mechanisms.

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
   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 [2], using the following

   l    left
   r    right
   c    center
   S    surround
   F    front
   R    rear

   channels    description     channel
                                  1       2     3     4     5     6

Schulzrinne                                                   [Page 7]

Internet Draft                  Profile                November 18, 1998

   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

   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

Schulzrinne                                                   [Page 8]

Internet Draft                  Profile                November 18, 1998

   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
   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

Schulzrinne                                                   [Page 9]

Internet Draft                  Profile                November 18, 1998

   (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

   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
   QCELP       frame           N/A               8,000          20
   SX7300P     frame           N/A               8,000          15           30
   SX8300P     frame           N/A               8,000          15           30
   SX9600P     frame           N/A               8,000          15           30
   VDVI        sample          var.               var.          20

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

   The characteristics of standard audio encodings are shown in Table 1;
   they are listed in order of their payload type in Table 4.  Entries
   with payload type "dyn" have a dynamic rather than static payload
   type. 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

Schulzrinne                                                  [Page 10]

Internet Draft                  Profile                November 18, 1998

   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.

   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.)

      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

Schulzrinne                                                  [Page 11]

Internet Draft                  Profile                November 18, 1998

   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 [7] as the IMA ADPCM wave

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

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

       o IMA ADPCM blocks contain an odd number of samples, since the
         first sample of a block is contained just in the header
         (uncompressed), followed by an even number of compressed
         samples. DVI4 has an even number of compressed samples only,
         using the 'predict' word from the header to decode the first

       o For DVI4, the 4-bit samples are packed with the first sample
         in the four most significant bits and the second sample in the
         four least significant bits. In the IMA ADPCM codec, the
         samples are packed in 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

Schulzrinne                                                  [Page 12]

Internet Draft                  Profile                November 18, 1998

   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
   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

Schulzrinne                                                  [Page 13]

Internet Draft                  Profile                November 18, 1998

   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
   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".

Schulzrinne                                                  [Page 14]

Internet Draft                  Profile                November 18, 1998

   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 ---------------->

   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 [8].

   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,

Schulzrinne                                                  [Page 15]

Internet Draft                  Profile                November 18, 1998

   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

   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|       |     |       |         |               |


Schulzrinne                                                  [Page 16]

Internet Draft                  Profile                November 18, 1998

   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.

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 [9,10,11]. 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.

Schulzrinne                                                  [Page 17]

Internet Draft                  Profile                November 18, 1998

   The order of the fields in the frame is described in Table 2. 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 the
   bit pattern described in Table 3.

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 -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.

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 2250 [12].

   Sampling rate and channel count are contained in the payload. MPEG-I
   audio supports sampling rates of 32, 44.1, and 48 kHz (ISO/IEC
   11172-3, section 1.1; "Scope"). MPEG-II additionally supports
   sampling rates of 16, 22.05 and 24 kHz.

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

Schulzrinne                                                  [Page 18]

Internet Draft                  Profile                November 18, 1998

        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

   given by Jayant and Noll [13].  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

Schulzrinne                                                  [Page 19]

Internet Draft                  Profile                November 18, 1998

   Octet     Bit 0      Bit 1      Bit 2      Bit 3      Bit 4      Bit 5      Bit 6      Bit 7
       0       1          1          0          1       LARc0.0    LARc0.1    LARc0.2    LARc0.3
       1    LARc0.4    LARc0.5    LARc1.0    LARc1.1    LARc1.2    LARc1.3    LARc1.4    LARc1.5
       2    LARc2.0    LARc2.1    LARc2.2    LARc2.3    LARc2.4    LARc3.0    LARc3.1    LARc3.2
       3    LARc3.3    LARc3.4    LARc4.0    LARc4.1    LARc4.2    LARc4.3    LARc5.0    LARc5.1
       4    LARc5.2    LARc5.3    LARc6.0    LARc6.1    LARc6.2    LARc7.0    LARc7.1    LARc7.2
       5     Nc0.0      Nc0.1      Nc0.2      Nc0.3      Nc0.4      Nc0.5      Nc0.6     bc0.0
       6     bc0.1      Mc0.0      Mc0.1     xmaxc00    xmaxc01    xmaxc02    xmaxc03    xmaxc04
       7    xmaxc05    xmc0.0     xmc0.1     xmc0.2     xmc1.0     xmc1.1     xmc1.2     xmc2.0
       8    xmc2.1     xmc2.2     xmc3.0     xmc3.1     xmc3.2     xmc4.0     xmc4.1     xmc4.2
       9    xmc5.0     xmc5.1     xmc5.2     xmc6.0     xmc6.1     xmc6.2     xmc7.0     xmc7.1
      10    xmc7.2     xmc8.0     xmc8.1     xmc8.2     xmc9.0     xmc9.1     xmc9.2     xmc10.0
      11    xmc10.1    xmc10.2    xmc11.0    xmc11.1    xmc11.2    xmc12.0    xmc12.1    xcm12.2
      12     Nc1.0      Nc1.1      Nc1.2      Nc1.3      Nc1.4      Nc1.5      Nc1.6      bc1.0
      13     bc1.1      Mc1.0      Mc1.1     xmaxc10    xmaxc11    xmaxc12    xmaxc13    xmaxc14
      14    xmax15     xmc13.0    xmc13.1    xmc13.2    xmc14.0    xmc14.1    xmc14.2    xmc15.0
      15    xmc15.1    xmc15.2    xmc16.0    xmc16.1    xmc16.2    xmc17.0    xmc17.1    xmc17.2
      16    xmc18.0    xmc18.1    xmc18.2    xmc19.0    xmc19.1    xmc19.2    xmc20.0    xmc20.1
      17    xmc20.2    xmc21.0    xmc21.1    xmc21.2    xmc22.0    xmc22.1    xmc22.2    xmc23.0
      18    xmc23.1    xmc23.2    xmc24.0    xmc24.1    xmc24.2    xmc25.0    xmc25.1    xmc25.2
      19     Nc2.0      Nc2.1      Nc2.2      Nc2.3      Nc2.4      Nc2.5      Nc2.6      bc2.0
      20     bc2.1      Mc2.0      Mc2.1     xmaxc20    xmaxc21    xmaxc22    xmaxc23    xmaxc24
      21    xmaxc25    xmc26.0    xmc26.1    xmc26.2    xmc27.0    xmc27.1    xmc27.2    xmc28.0
      22    xmc28.1    xmc28.2    xmc29.0    xmc29.1    xmc29.2    xmc30.0    xmc30.1    xmc30.2
      23    xmc31.0    xmc31.1    xmc31.2    xmc32.0    xmc32.1    xmc32.2    xmc33.0    xmc33.1
      24    xmc33.2    xmc34.0    xmc34.1    xmc34.2    xmc35.0    xmc35.1    xmc35.2    xmc36.0
      25    Xmc36.1    xmc36.2    xmc37.0    xmc37.1    xmc37.2    xmc38.0    xmc38.1    xmc38.2
      26     Nc3.0      Nc3.1      Nc3.2      Nc3.3      Nc3.4      Nc3.5      Nc3.6      bc3.0
      27     bc3.1      Mc3.0      Mc3.1     xmaxc30    xmaxc31    xmaxc32    xmaxc33    xmaxc34
      28    xmaxc35    xmc39.0    xmc39.1    xmc39.2    xmc40.0    xmc40.1    xmc40.2    xmc41.0
      29    xmc41.1    xmc41.2    xmc42.0    xmc42.1    xmc42.2    xmc43.0    xmc43.1    xmc43.2
      30    xmc44.0    xmc44.1    xmc44.2    xmc45.0    xmc45.1    xmc45.2    xmc46.0    xmc46.1
      31    xmc46.2    xmc47.0    xmc47.1    xmc47.2    xmc48.0    xmc48.1    xmc48.2    xmc49.0
      32    xmc49.1    xmc49.2    xmc50.0    xmc50.1    xmc50.2    xmc51.0    xmc51.1    xmc51.2

   Table 3: GSM payload format

   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 QCELP

   The packetization of the QCELP audio codec is described in [14].

Schulzrinne                                                  [Page 20]

Internet Draft                  Profile                November 18, 1998

4.5.17 RED

   The redundant audio payload format "RED" is specified by RFC 2198
   [15]. 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 SX*

   The SX7300P, SX8300P and SX9600P codecs are part of the same
   compatible family and distinguished by the first octet in each frame,
   where "x" can be any value:

      0 1 2 3 4 5 6 7
     |0 0 x          |  SX7300P bitstream (14 byte frame)
     |0 1 0          |  SX8300P bitstream (16 byte frame)
     |1 0 x          |  VAD bistream      ( 2 byte frame)
     |1 1 x          |  SX9600P bitstream (18 byte frame)
     +-+-+-+-+-+-+-+-+ 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. 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
   ms) into an encoded frame of 16 octets, yielding an encoded bit rate
   of approximately 8533 b/s. SX9600P

   The SX9600P is a low-complexity, toll-quality 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 18 octets, yielding an

Schulzrinne                                                  [Page 21]

Internet Draft                  Profile                November 18, 1998

   encoded bit rate of 9600 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 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
                                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 [16].

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 [17].

5.3 H261

Schulzrinne                                                  [Page 22]

Internet Draft                  Profile                November 18, 1998

   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 [18].

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 [19].

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 2250
   [12], 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 2250 [12],
   Section 2.

5.7 MP1S

   MP1S designates an MPEG-I systems stream, encapsulated according to
   RFC 2250 [12].

5.8 MP2P

   MP2P designates an MPEG-II program stream, encapsulated according to
   RFC 2250 [12].

5.9 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

Schulzrinne                                                  [Page 23]

Internet Draft                  Profile                November 18, 1998

   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. In addition, payload type
   values in the range 96-127 may be defined dynamically through a
   conference control protocol, which is beyond the scope of this
   document. For example, a session directory could specify that for a
   given session, payload type 96 indicates PCMU encoding, 8,000 Hz
   sampling rate, 2 channels. 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.

   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.

Schulzrinne                                                  [Page 24]

Internet Draft                  Profile                November 18, 1998

   The standard video encodings and their payload types are listed in
   Table 4.

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 [20] 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] M. Handley and V. Jacobson, "SDP: Session Description Protocol,"
   Request for Comments (Proposed Standard) RFC 2327, Internet
   Engineering Task Force, Apr. 1998.

Schulzrinne                                                  [Page 25]

Internet Draft                  Profile                November 18, 1998

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

   [3] 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.

   [4] 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.

   [5] 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.

   [6] 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.

   [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] D. Deleam 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.

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

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

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

   [12] D. Hoffman, G. Fernando, V. Goyal, and M. Civanlar, "RTP payload
   format for MPEG1/MPEG2 video," Request for Comments (Proposed
   Standard) RFC 2250, Internet Engineering Task Force, Jan. 1998.

   [13] N. S. Jayant and P. Noll, Digital Coding of Waveforms--
   Principles and Applications to Speech and Video Englewood Cliffs, New

Schulzrinne                                                  [Page 26]

Internet Draft                  Profile                November 18, 1998

      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         QCELP         A             8000          1
      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--95     unassigned    ?
      96--127    dynamic       ?
      dyn        RED           A
      dyn        MP1S          V             90000
      dyn        MP2P          V             90000

   Table 4: Payload types (PT) for standard audio and video encodings

Schulzrinne                                                  [Page 27]

Internet Draft                  Profile                November 18, 1998

   Jersey: Prentice-Hall, 1984.

   [14] K. McKay, "RTP Payload Format for PureVoice(tm) Audio", Internet
   Draft, Internet Engineering Task Force, Oct. 1998.  Work in progress.

   [15] C. Perkins, I. Kouvelas, O. Hodson, V. Hardman, M. Handley, J.C.
   Bolot, A. Vega-Garcia, and S. Fosse-Parisis, "RTP Payload for
   Redundant Audio Data," Request for Comments (Proposed Standard) RFC
   2198, Internet Engineering Task Force, Sep. 1997.

   [16] 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.

   [17] 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.

   [18] 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.

   [19] C. Zhu, "RTP payload format for H.263 video streams," Request
   for Comments (Proposed Standard) RFC 2190, Internet Engineering Task
   Force, Sep. 1997.

   [20] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming
   protocol (RTSP)," Request for Comments (Proposed Standard) RFC 2326,
   Internet Engineering Task Force, Apr. 1998.

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:

Schulzrinne                                                  [Page 28]

Internet Draft                  Profile                November 18, 1998

   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



   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

Schulzrinne                                                  [Page 29]

Internet Draft                  Profile                November 18, 1998

   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.

   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

Schulzrinne                                                  [Page 30]

Internet Draft                  Profile                November 18, 1998


   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 [7].

                           Table of Contents

   1          Introduction ........................................    2
   2          RTP and RTCP Packet Forms and Protocol Behavior .....    3
   3          Registering Additional Encodings with IANA ..........    5
   4          Audio ...............................................    7
   4.1        Encoding-Independent Rules ..........................    7
   4.2        Operating Recommendations ...........................    8
   4.3        Guidelines for Sample-Based Audio Encodings .........    8
   4.4        Guidelines for Frame-Based Audio Encodings ..........    9
   4.5        Audio Encodings .....................................   10
   4.5.1      1016 ................................................   11
   4.5.2      CN ..................................................   11
   4.5.3      DVI4 ................................................   12
   4.5.4      G722 ................................................   13
   4.5.5      G723 ................................................   13
   4.5.6      G726-16, G726-24, G726-32, G726-40 ..................   14
   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 ......................   18
   4.5.11     L8 ..................................................   18
   4.5.12     L16 .................................................   18
   4.5.13     LPC .................................................   18
   4.5.14     MPA .................................................   18
   4.5.15     PCMA and PCMU .......................................   18
   4.5.16     QCELP ...............................................   20
   4.5.17     RED .................................................   21

Schulzrinne                                                  [Page 31]

Internet Draft                  Profile                November 18, 1998

   4.5.18     SX* .................................................   21   SX7300P .............................................   21   SX8300P .............................................   21   SX9600P .............................................   21
   4.5.19     VDVI ................................................   22
   5          Video ...............................................   22
   5.1        CelB ................................................   22
   5.2        JPEG ................................................   22
   5.3        H261 ................................................   22
   5.4        H263 ................................................   23
   5.5        MPV .................................................   23
   5.6        MP2T ................................................   23
   5.7        MP1S ................................................   23
   5.8        MP2P ................................................   23
   5.9        nv ..................................................   23
   6          Payload Type Definitions ............................   23
   7          RTP over TCP and Similar Byte Stream Protocols ......   25
   8          Port Assignment .....................................   25
   9          Bibliography ........................................   25
   10         Acknowledgements ....................................   28
   11         Address of Author ...................................   28

Schulzrinne                                                  [Page 32]