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Versions: 00 01 02 03 04 05 06 07 08 09 10 11 12 RFC 3551

Internet Engineering Task Force                                   AVT WG
Internet Draft                                        Schulzrinne/Casner
draft-ietf-avt-profile-new-07.txt                    Columbia U./Cisco Systems
October 21, 1999
Expires: April 21, 2000

    RTP Profile for Audio and Video Conferences with Minimal Control


   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   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-

   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".

   The list of current Internet-Drafts can be accessed at

   The list of Internet-Draft Shadow Directories can be accessed at



   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.

   This document describes a profile called "RTP/AVP" for the use of the
   real-time transport protocol (RTP), version 2, and the associated
   control protocol, RTCP, within audio and video multiparticipant
   conferences with minimal control. It provides interpretations of
   generic fields within the RTP specification suitable for audio and
   video conferences. In particular, this document defines a set of
   default mappings from payload type numbers to encodings.

   This document also describes how audio and video data may be carried
   within RTP. It defines a set of standard encodings and their names

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   when used within RTP. The descriptions provide pointers to reference
   implementations and the detailed standards. This document is meant as
   an aid for implementors of audio, video and other real-time
   multimedia applications.

   Resolution of Open Issues

   [Note to the RFC Editor: This section is to be deleted when this
   draft is published as an RFC but is shown here for reference during
   the Last Call. The first paragraph of the Abstract is also to be
   deleted.  All RFC XXXX should be filled in with the number of the RTP
   specification RFC submitted for Draft Standard status, and all RFC
   YYYY should be filled in with the number of the draft specifying MIME
   registration of RTP payload types as it is submitted for Proposed
   Standard status. These latter references are intended to be non-

   Readers are directed to Appendix 9, Changes from RFC 1890, for a
   listing of the changes that have been made in this draft.  The
   changes from RFC 1890 are marked with change bars in the PostScript
   form of this draft.

   The revisions in this draft are intended to be complete for Last
   Call.  The following open issues from previous drafts have been

       o  The procedure for registering RTP encoding names as MIME
          subtypes was moved to a separate RFC-to-be that may also serve
          to specify how (some of) the encodings here may be used with
          mail and other not-RTP transports. That procedure is not
          required to implement this profile, but may be used in those
          contexts where it is needed.

       o  This profile follows the suggestion in the RTP spec that RTCP
          bandwidth may be specified separately from the session
          bandwidth and separately for active senders and passive

       o  No specific action is taken in this document to address
          generic payload formats; it is assumed that if any generic
          payload formats are developed, they can be specified in
          separate RFCs and that the session parameters they require for
          operation can be specified in the MIME registration of those

1 Introduction

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   This profile defines aspects of RTP left unspecified in the RTP
   Version 2 protocol definition (RFC XXXX) [1].  This profile is
   intended for the use within audio and video conferences with minimal
   session control. In particular, no support for the negotiation of
   parameters or membership control is provided. The profile is expected
   to be useful in sessions where no negotiation or membership control
   are used (e.g., using the static payload types and the membership
   indications provided by RTCP), but this profile may also be useful in
   conjunction with a higher-level control protocol.

   Use of this profile may be implicit in the use of the appropriate
   applications; there may be no explicit indication by port number,
   protocol identifier or the like.  Applications such as session
   directories 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 encodings and payload formats for
   audio and video.

1.1 Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in RFC 2119 [2] and
   indicate requirement levels for implementations compliant with this
   RTP profile.

   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.

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        RTP data header extensions: No RTP header extensions are
             defined, but applications operating under this profile MAY
             use such extensions. Thus, applications SHOULD NOT assume
             that the RTP header X bit is always zero and SHOULD be
             prepared to ignore the header extension. If a header
             extension is defined in the future, that definition MUST
             specify the contents of the first 16 bits in such a way
             that multiple different extensions can be identified.

        RTCP packet types: No additional RTCP packet types are defined
             by this profile specification.

        RTCP report interval: The suggested constants are to be used for
             the RTCP report interval calculation.  Sessions operating
             under this profile MAY specify a separate parameter for the
             RTCP traffic bandwidth rather than using the default
             fraction of the session bandwidth. The RTCP traffic
             bandwidth MAY be divided into two separate session
             parameters for those participants which are active data
             senders and those which are not. Following the
             recommendation in the RTP specification [1] that 1/4 of the
             RTCP bandwidth be dedicated to data senders, the
             RECOMMENDED default values for these two parameters would
             be 1.25% and 3.75%, respectively. For a particular session,
             the RTCP bandwidth for non-data-senders MAY be set to zero
             when operating on unidirectional links or for sessions that
             don't require feedback on the quality of reception. The
             RTCP bandwidth for data senders SHOULD be kept non-zero so
             that sender reports can still be sent for inter-media
             synchronization and to identify the source by CNAME. The
             means by which the one or two session parameters for RTCP
             bandwidth are specified is beyond the scope of this memo.

        SR/RR extension: No extension section is defined for the RTCP SR
             or RR packet.

        SDES use: Applications MAY use any of the SDES items described
             in the RTP specification. While CNAME information MUST be
             sent every reporting interval, other items SHOULD only be
             sent every third reporting interval, with NAME sent seven
             out of eight times within that slot and the remaining SDES
             items cyclically taking up the eighth slot, as defined in
             Section 6.2.2 of the RTP specification. In other words,
             NAME is sent in RTCP packets 1, 4, 7, 10, 13, 16, 19,
             while, say, EMAIL is used in RTCP packet 22.

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

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        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

             It is RECOMMENDED that pass phrases be 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 email.

             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 `white space' to be the ASCII space,
             formfeed, newline, carriage 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

        Underlying protocol: The profile specifies the use of RTP over
             unicast and multicast UDP as well as TCP.  (This does not

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             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: A minimal TCP encapsulation is defined.

3 Registering Additional Encodings with IANA

   This profile lists a set of encodings, each of which is comprised of
   a particular media data compression or representation plus a payload
   format for encapsulation within RTP. Some of those payload formats
   are specified here, while others are specified in separate RFCs. It
   is expected that additional encodings beyond the set listed here will
   be created in the future and specified in additional payload format

   This profile also assigns to each encoding a short name which MAY be
   used by higher-level control protocols, such as the Session
   Description Protocol (SDP), RFC 2327 [5], to identify encodings
   selected for a particular RTP session.

   In some contexts it may be useful to refer to these encodings in the
   form of a MIME content-type. To facilitate this, RFC YYYY [3]
   provides registrations for all of the encodings names listed here as
   MIME subtype names under the "audio" and "video" MIME types through
   the MIME registration procedure as specified in RFC 2048 [4].

   Any additional encodings specified for use under this profile (or
   others) may also be assigned names registered as MIME subtypes with
   the Internet Assigned Numbers Authority (IANA). This registry
   provides a means to insure that the names assigned to the additional
   encodings are kept unique. RFC YYYY specifies the information that is
   required for the registration of RTP encodings.

   In addition to assigning names to encodings, this profile also also
   assigns static RTP payload type numbers to some of them. However, the
   payload type number space is relatively small and cannot accommodate
   assignments for all existing and future encodings. During the early
   stages of RTP development, it was necessary to use statically
   assigned payload types because no other mechanism had been specified
   to bind encodings to payload types. It was anticipated that non-RTP
   means beyond the scope of this memo (such as directory services or
   invitation protocols) would be specified to establish a dynamic
   mapping between a payload type and an encoding. Now, mechanisms for
   defining dynamic payload type bindings have been specified in the
   Session Description Protocol (SDP) and in other protocols such as

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   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
   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, SHOULD be 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 MUST set the marker bit to

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   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 [6], using the following

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

   channels  description   channel
                              1     2   3   4   5   6
   2         stereo           l     r
   3                          l     r   c
   4         quadrophonic    Fl     Fr  Rl  Rr
   4                          l     c   r   S
   5                         Fl     Fr  Fc  Sl  Sr
   6                          l     lc  c   r   rc  S

   Samples for all channels belonging to a single sampling instant MUST
   be within the same packet. The interleaving of samples from different
   channels depends on the encoding. General guidelines are given in
   Section 4.3 and 4.4.

   The sampling frequency SHOULD be drawn from the set: 8000, 11025,
   16000, 22050, 24000, 32000, 44100 and 48000 Hz.  (Older Apple
   Macintosh computers had a native sample rate of 22254.54 Hz, which
   can be converted to 22050 with acceptable quality by dropping 4
   samples in a 20 ms frame.)  However, most audio encodings are defined
   for a more restricted set of sampling frequencies. Receivers SHOULD
   be prepared to accept multi-channel audio, but MAY choose to only

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   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
   for links with severe bandwidth constraints, a higher packetization
   delay MAY be used.  A receiver SHOULD accept packets representing
   between 0 and 200 ms of audio data. (For framed audio encodings, a
   receiver SHOULD accept packets with a number of frames equal to 200
   ms divided by the frame duration, rounded up.) This restriction
   allows reasonable buffer sizing for the receiver.

4.3 Guidelines for Sample-Based Audio Encodings

   In sample-based encodings, each audio sample is represented by a
   fixed number of bits. Within the compressed audio data, codes for
   individual samples may span octet boundaries. An RTP audio packet may
   contain any number of audio samples, subject to the constraint that
   the number of bits per sample times the number of samples per packet
   yields an integral octet count. Fractional encodings produce less
   than one octet per sample.

   The duration of an audio packet is determined by the number of
   samples in the packet.

   For sample-based encodings producing one or more octets per sample,
   samples from different channels sampled at the same sampling instant
   SHOULD be packed in consecutive octets. For example, for a two-
   channel encoding, the octet sequence is (left channel, first sample),
   (right channel, first sample), (left channel, second sample), (right
   channel, second sample), .... For multi-octet encodings, octets
   SHOULD be transmitted in network byte order (i.e., most significant
   octet first).

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

   The RTP timestamp reflects the instant at which the first sample in
   the packet was sampled, that is, the oldest information in the

4.4 Guidelines for Frame-Based Audio Encodings

   Frame-based encodings encode a fixed-length block of audio into
   another block of compressed data, typically also of fixed length. For
   frame-based encodings, the sender MAY choose to combine several such
   frames into a single RTP packet. The receiver can tell the number of
   frames contained in an RTP packet, if all the frames have the same
   length, by dividing the RTP payload length by the audio frame size
   which is defined as part of the encoding. This does not work when
   carrying frames of different sizes unless the frame sizes are
   relatively prime.  If not, the frames MUST indicate their size.

   For frame-based codecs, the channel order is defined for the whole
   block. That is, for two-channel audio, right and left samples SHOULD
   be coded independently, with the encoded frame for the left channel
   preceding that for the right channel.

   All frame-oriented audio codecs SHOULD be able to encode and decode
   several consecutive frames within a single packet. Since the frame
   size for the frame-oriented codecs is given, there is no need to use
   a separate designation for the same encoding, but with different
   number of frames per packet.

   RTP packets SHALL contain a whole number of frames, with frames
   inserted according to age within a packet, so that the oldest frame
   (to be played first) occurs immediately after the RTP packet header.
   The RTP timestamp reflects the instant at which the first sample in
   the first frame was sampled, that is, the oldest information in the

4.5 Audio Encodings

   The characteristics of the audio encodings described in this document
   are shown in Table 1; they are listed in order of their payload type
   in Table 4.  While most audio codecs are only specified for a fixed
   sampling rate, some sample-based algorithms (indicated by an entry of
   "var." in the sampling rate column of Table 1) may be used with
   different sampling rates, resulting in different coded bit rates.
   When used with a sampling rate other than that for which a static
   payload type is defined, non-RTP means beyond the scope of this memo

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    name of                              sampling              default
    encoding  sample/frame  bits/sample      rate  ms/frame  ms/packet
    1016      frame         N/A             8,000        30         30
    CN        frame         N/A              var.
    DVI4      sample        4                var.                   20
    G722      sample        8              16,000                   20
    G723      frame         N/A             8,000        30         30
    G726-32   sample        4               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
    GSM-HR    frame         N/A             8,000        20         20
    GSM-EFR   frame         N/A             8,000        20         20
    L8        sample        8                var.                   20
    L16       sample        16               var.                   20
    LPC       frame         N/A             8,000        20         20
    MPA       frame         N/A              var.      var.
    PCMA      sample        8                var.                   20
    PCMU      sample        8                var.                   20
    QCELP     frame         N/A             8,000        20         20
    VDVI      sample        var.             var.                   20

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

   MUST be used to define a dynamic payload type and MUST indicate the
   selected RTP timestamp clock rate, which is usually the same as the
   sampling rate for audio.

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 the rate of noise level updates is implementation specific. The
   magnitude of the noise level is packed into the least significant
   bits of the noise-level payload, as shown below.

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   The noise level is expressed in -dBov, with values from 0 to 127
   representing 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.) For example, in a 16-bit linear
   PCM system (L16), a signal with 0 dBov represents a square wave with
   the maximum possible amplitude (+/-32767), and -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 MUST be defined through dynamic
   payload types. The RTP packet SHOULD NOT have the marker bit set.

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

4.5.3 DVI4

   DVI4 is specified, with pseudo-code, in [11] 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

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

   Each packet contains a single DVI block. This profile only defines
   the 4-bit-per-sample version, while IMA also 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 because there is no means
   to indicate a partially filled last octet.

   Packing of samples for multiple channels is for further study.

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

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

4.5.4 G722

   G722 is specified in ITU-T Recommendation G.722, "7 kHz audio-coding
   within 64 kbit/s".  The G.722 encoder produces a stream of octets,
   each of which SHALL be octet-aligned in an RTP packet. The first bit
   transmitted in the G.722 octet, which is the most significant bit of
   the higher sub-band sample, SHALL correspond to the most significant
   bit of the octet in the RTP packet.

   Even though the actual sampling rate for G.722 audio is 16000 Hz, the
   RTP clock rate for the G722 payload format is 8000 Hz because that
   value was erroneously assigned in RFC 1890 and must remain unchanged

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   for backward compatibility. The octet rate or sample-pair rate is
   8000 Hz.

4.5.5 G723

   G723 is specified in ITU Recommendation G.723.1, "Dual-rate speech
   coder for multimedia communications transmitting at 5.3 and 6.3
   kbit/s". The G.723.1 5.3/6.3 kbit/s codec was defined by the ITU-T as
   a mandatory codec for ITU-T H.324 GSTN videophone terminal
   applications.  The algorithm has a floating point specification in
   Annex B to G.723.1, a silence compression algorithm in Annex A to
   G.723.1 and 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

   The packing of the encoded bit stream into octets and the
   transmission order of the octets is specified in G.723.1.

4.5.6 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

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   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).
   Packetization is specified here only for the 32 kb/s encoding which
   is labeled G726-32.

   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 payload-specific 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.  The number of samples per packet
   MUST be even because there is no means to indicate a partially filled
   last octet.

4.5.7 G728

   G728 is specified in ITU-T Recommendation G.728, "Coding of speech at
   16 kbit/s using low-delay code excited linear prediction".

   A G.278 encoder translates 5 consecutive audio samples into a 10-bit
   codebook index, resulting in a bit rate of 16 kb/s for audio sampled
   at 8,000 samples per second. The group of five consecutive samples is
   called a vector. Four consecutive vectors, labeled V1 to V4 (where V1
   is to be played first by the receiver), build one G.728 frame. The
   four vectors of 40 bits are packed into 5 octets, labeled B1 through
   B5. B1 SHALL be placed first in the RTP packet.

   Referring to the figure below, the principle for bit order is
   "maintenance of bit significance". Bits from an older vector are more
   significant than bits from newer vectors. The MSB of the frame goes
   to the MSB of B1 and the LSB of the frame goes to LSB of B5.

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

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   <------------- 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.

4.5.8 G729

   G729 is specified in ITU-T Recommendation G.729, "Coding of speech at
   8 kbit/s using conjugate structure-algebraic code excited linear
   prediction (CS-ACELP)". A reduced-complexity version of the G.729
   algorithm is specified in Annex A to Rec. G.729. The speech coding
   algorithms in the main body of G.729 and in G.729 Annex A are fully
   interoperable with each other, so there is no need to further
   distinguish between them. 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 [12].

   A voice activity detector (VAD) and comfort noise generator (CNG)
   algorithm in Annex B of G.729 is RECOMMENDED for digital simultaneous
   voice and data applications and can be used in conjunction with G.729
   or G.729 Annex A. A G.729 or G.729 Annex A frame contains 10 octets,
   while the G.729 Annex B comfort noise frame occupies 2 octets:

    0                   1
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
   |L|  LSF1   |  LSF2 |   GAIN  |R|
   |S|         |       |         |E|
   |F|0 1 2 3 4|0 1 2 3|0 1 2 3 4|S|
   |0|         |       |         |V|    RESV = Reserved (zero)

   An RTP packet may consist of zero or more G.729 or G.729 Annex A
   frames, followed by zero or one G.729 Annex B payloads. The presence
   of a comfort noise frame can be deduced from the length of the RTP

   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.

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   The mapping of the these parameters is given below. Bits are numbered
   as Internet order, that is, the most significant bit is bit 0.

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   |L|      L1     |    L2   |    L3   |       P1      |P|    C1   |
   |0|             |         |         |               |0|         |
   | |0 1 2 3 4 5 6|0 1 2 3 4|0 1 2 3 4|0 1 2 3 4 5 6 7| |0 1 2 3 4|
   | |             |         |         |               | |         |

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

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

4.5.9 GSM

   GSM (group speciale mobile) denotes the European GSM 06.10 standard
   for full-rate speech transcoding, ETS 300 961, which is based on
   RPE/LTP (residual pulse excitation/long term prediction) coding at a
   rate of 13 kb/s [13,14,15]. The text of the standard can be obtained

   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

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

   In the GSM packing used by RTP, the bits SHALL be packed beginning
   from the most significant bit. Every 160 sample GSM frame is coded
   into one 33 octet (264 bit) buffer. Every such buffer begins with a 4
   bit signature (0xD), followed by the MSB encoding of the fields of
   the frame. The first octet thus contains 1101 in the 4 most
   significant bits (0-3) and the 4 most significant bits of F1 (0-3) in
   the 4 least significant bits (4-7). The second octet contains the 2
   least significant bits of F1 in bits 0-1, and F2 in bits 2-7, and so
   on. The order of the fields in the frame is described in Table 2. GSM variable names and numbers

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

4.5.10 GSM-HR

   GSM-HR denotes GSM 06.20 half rate speech transcoding, specified in
   ETS 300 969 which is available from ETSI at the address given in
   Section 4.5.9. This codec has a frame length of 112 bits (14 octets).
   Packing of the fields in the codec bit stream into octets for
   transmission in RTP is done in a manner similar to that specified
   here for the original GSM 06.10 codec and is specified in ETSI
   Technical Specification TS 101 318.

4.5.11 GSM-EFR

   GSM-EFR denotes GSM 06.60 enhanced full rate speech transcoding,
   specified in ETS 300 969 which is available from ETSI at the address
   given in Section 4.5.9. This codec has a frame length of 244 bits.
   For transmission in RTP, each codec frame is packed into a 31 octet
   (248 bit) buffer beginning with a 4-bit signature 0xC in a manner

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

   Table 2: Ordering of GSM variables

   similar to that specified here for the original GSM 06.10 codec. The
   packing is specified in ETSI Technical Specification TS 101 318.

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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

   Table 3: GSM payload format

4.5.12 L8

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

4.5.13 L16

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   L16 denotes uncompressed audio data samples, 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 transmitted in network
   byte order (most significant byte first).

4.5.14 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.

4.5.15 MPA

   MPA denotes MPEG-1 or MPEG-2 audio encapsulated as elementary
   streams.  The encoding is defined in ISO standards ISO/IEC 11172-3
   and 13818-3.  The encapsulation is specified in RFC 2250 [16].

   The encoding may be at any of three levels of complexity, called
   Layer I, II and III. The selected layer as well as the sampling rate
   and channel count are indicated in the payload. The RTP timestamp
   clock rate is always 90000, independent of the sampling rate.  MPEG-1
   audio supports sampling rates of 32, 44.1, and 48 kHz (ISO/IEC
   11172-3, section 1.1; "Scope"). MPEG-2 supports sampling rates of 16,
   22.05 and 24 kHz.  The number of samples per frame is fixed, but the
   frame size will vary with the sampling rate and bit rate.

4.5.16 PCMA and PCMU

   PCMA and PCMU are specified in ITU-T Recommendation G.711. Audio data
   is encoded as eight bits per sample, after logarithmic scaling. PCMU
   denotes mu-law scaling, PCMA A-law scaling. A detailed description is
   given by Jayant and Noll [17].  Each G.711 octet SHALL be octet-
   aligned in an RTP packet. The sign bit of each G.711 octet SHALL
   correspond to the most significant bit of the octet in the RTP packet
   (i.e., assuming the G.711 samples are handled as octets on the host
   machine, the sign bit SHALL be the most signficant bit of the octet
   as defined by the host machine format). The 56 kb/s and 48 kb/s modes
   of G.711 are not applicable to RTP, since PCMA and PCMU SHALL always
   be transmitted as 8-bit samples.

4.5.17 QCELP

   The Electronic Industries Association (EIA) & Telecommunications
   Industry Association (TIA) standard IS-733, "TR45: High Rate Speech

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   Service Option for Wideband Spread Spectrum Communications Systems,"
   defines the QCELP audio compression algorithm for use in wireless
   CDMA applications. The QCELP CODEC compresses each 20 milliseconds of
   8000 Hz, 16- bit sampled input speech into one of four different size
   output frames: Rate 1 (266 bits), Rate 1/2 (124 bits), Rate 1/4 (54
   bits) or Rate 1/8 (20 bits). For typical speech patterns, this
   results in an average output of 6.8 k bits/sec for normal mode and
   4.7 k bits/sec for reduced rate mode. The packetization of the QCELP
   audio codec is described in [18].

4.5.18 RED

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

4.5.19 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.  The last octet is padded with 1 bits if the last
   sample does not fill the last octet. This padding is distinct from
   the valid codewords.  The receiver needs to detect the padding
   because there is no explicit count of samples in the packet.

   It uses the following encoding:

                      DVI4 codeword  VDVI bit pattern
                                  0  00
                                  1  010
                                  2  1100
                                  3  11100
                                  4  111100
                                  5  1111100
                                  6  11111100
                                  7  11111110
                                  8  10
                                  9  011
                                 10  1101

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                                 11  11101
                                 12  111101
                                 13  1111101
                                 14  11111101
                                 15  11111111

5 Video

   The following sections describe the video encodings that are defined
   in this memo and give their abbreviated names used for
   identification.  These video encodings and their payload types are
   listed in Table 5.

   All of these video encodings use an RTP timestamp frequency of 90,000
   Hz, the same as the MPEG presentation time stamp frequency. This
   frequency yields exact integer timestamp increments for the typical
   24 (HDTV), 25 (PAL), and 29.97 (NTSC) and 30 Hz (HDTV) frame rates
   and 50, 59.94 and 60 Hz field rates. While 90 kHz is the RECOMMENDED
   rate for future video encodings used within this profile, other rates
   MAY be used.  However, it is not sufficient to use the video frame
   rate (typically between 15 and 30 Hz) because that does not provide
   adequate resolution for typical synchronization requirements when
   calculating the RTP timestamp corresponding to the NTP timestamp in
   an RTCP SR packet. The timestamp resolution MUST also be sufficient
   for the jitter estimate contained in the receiver reports.

   For most of these video encodings, the RTP timestamp encodes the
   sampling instant of the video image contained in the RTP data packet.
   If a video image occupies more than one packet, the timestamp is the
   same on all of those packets. Packets from different video images are
   distinguished by their different timestamps.

   Most of these video encodings also specify that the marker bit of the
   RTP header SHOULD be set to one in the last packet of a video frame
   and otherwise set to zero. Thus, it is not necessary to wait for a
   following packet with a different timestamp to detect that a new
   frame should be displayed.

5.1 BT656

   The encoding is specified in ITU-R Recommendation BT.656-3,
   "Interfaces for Digital Component Video Signals in 525-Line and 625-
   Line Television Systems operating at the 4:2:2 Level of
   Recommendation ITU-R BT.601 (Part A)". The packetization and RTP-
   specific properties are described in RFC 2431 [20].

5.2 CelB

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   The CELL-B encoding is a proprietary encoding proposed by Sun
   Microsystems. The byte stream format is described in RFC 2029 [21].

5.3 JPEG

   The encoding is specified in ISO Standards 10918-1 and 10918-2. The
   RTP payload format is as specified in RFC 2435 [22].

5.4 H261

   The encoding is specified in ITU-T Recommendation H.261, "Video codec
   for audiovisual services at p x 64 kbit/s". The packetization and
   RTP-specific properties are described in RFC 2032 [23].

5.5 H263

   The encoding is specified in the 1996 version of ITU-T Recommendation
   H.263, "Video coding for low bit rate communication". The
   packetization and RTP-specific properties are described in RFC 2190

5.6 H263-1998

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

5.7 MPV

   MPV designates the use of MPEG-1 and MPEG-2 video encoding elementary
   streams as specified in ISO Standards ISO/IEC 11172 and 13818-2,
   respectively. The RTP payload format is as specified in RFC 2250
   [16], Section 3.

5.8 MP2T

   MP2T designates the use of MPEG-2 transport streams, for either audio
   or video. The RTP payoad format is described in RFC 2250 [16],
   Section 2.

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5.9 MP1S

   MP1S designates an MPEG-1 systems stream, encapsulated according to
   RFC 2250 [16].

5.10 MP2P

   MP2P designates an MPEG-2 program stream, encapsulated according to
   RFC 2250 [16].

5.11 BMPEG

   BMPEG designates an experimental payload format for MPEG-1 and MPEG-2
   which specifies bundled (multiplexed) transport of audio and video
   elementary streams in one RTP stream as an alternative to the MP1S
   and MP2P formats. The packetization is described in RFC 2343 [26].

5.12 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: frederic@parc.xerox.com

6 Payload Type Definitions

   Tables 4 and 5 define this profile's static payload type values for
   the PT field of the RTP data header.  In addition, payload type
   values in the range 96-127 MAY be defined dynamically through a
   conference control protocol, which is beyond the scope of this
   document. For example, a session directory could specify that for a
   given session, payload type 96 indicates PCMU encoding, 8,000 Hz
   sampling rate, 2 channels.  Entries in Tables 4 and 5 with payload
   type "dyn" have no static payload type assigned and are only used
   with a dynamic payload type. 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).

   The payload types currently defined in this profile are assigned to
   exactly one of three categories or media types : audio only, video
   only and those combining audio and video. The media types are marked

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   in Tables 4 and 5 as "A", "V" and "AV", respectively.  Payload types
   of different media types SHALL NOT be interleaved or multiplexed
   within a single RTP session, but multiple RTP sessions MAY be used in
   parallel to send multiple media types. An RTP source MAY change
   payload types within the same media type during a session.  See the
   section "Multiplexing RTP Sessions" of RFC XXXX for additional

   Session participants agree through mechanisms beyond the scope of
   this specification on the set of payload types allowed in a given
   session.  This set MAY, for example, be defined by the capabilities
   of the applications used, negotiated by a conference control protocol
   or established by agreement between the human participants.

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

7 RTP over TCP and Similar Byte Stream Protocols

   Under special circumstances, it may be necessary to carry RTP in
   protocols offering a byte stream abstraction, such as TCP, possibly
   multiplexed with other data. 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 [27] provides its own encapsulation and does not need an
   extra length indication.)

8 Port Assignment

   As specified in the RTP protocol definition, RTP data SHOULD be
   carried on an even UDP or TCP port number and the corresponding RTCP
   packets SHOULD 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

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            PT   encoding    media type  clock rate  channels
                 name                    (Hz)
            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           8000        1
            10   L16         A           44100       2
            11   L16         A           44100       1
            12   QCELP       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   unassigned  A           8000        1
            20   unassigned  A
            21   unassigned  A
            22   unassigned  A
            23   unassigned  A
            dyn  GSM-HR      A           8000        1
            dyn  GSM-EFR     A           8000        1
            dyn  RED         A

   Table 4: Payload types (PT) for audio encodings

   default pair. Applications that operate under multiple profiles MAY
   use this port pair as an indication to select this profile if they
   are not subject to the constraint of the previous paragraph.
   Applications need not have a default and MAY require that the port
   pair be explicitly specified. The particular port numbers were chosen
   to lie in the range above 5000 to accommodate port number allocation
   practice within some versions of the Unix operating system, where
   port numbers below 1024 can only be used by privileged processes and
   port numbers between 1024 and 5000 are automatically assigned by the
   operating system.

9 Changes from RFC 1890

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               PT      encoding    media type  clock rate
                       name                    (Hz)
               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
               77-95   unassigned  ?
               96-127  dynamic     ?
               dyn     BT656       V           90000
               dyn     H263-1998   V           90000
               dyn     MP1S        V           90000
               dyn     MP2P        V           90000
               dyn     BMPEG       V           90000

   Table 5: Payload types (PT) for video and combined encodings

   This RFC revises RFC 1890. It is fully backwards-compatible with RFC
   1890 and codifies existing practice. The changes are listed below.

       o  Additional payload formats and/or expanded descriptions were
          included for CN, G722, G723, G726, G728, G729, GSM, GSM-HR,
          GSM-EFR, QCELP, RED, VDVI, BT656, H263-1998, MP1S, MP2P and

       o  Static payload types 4, 12, 13, 16, 17, 18 and 34 were added.

       o  The policy is established that no additional registration of
          static payload types for this Profile will be made beyond
          those included in Tables 4 and 5, but additional encoding
          names may be registered as MIME subtypes.

       o  In Section 4.1, the requirement level for setting of the
          marker bit on the first packet after silence for audio was
          changed from "is" to "SHOULD be".

       o  Similarly, text was added to specify that the marker bit

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          SHOULD be set to one on the last packet of a video frame, and
          that video frames are distinguished by their timestamps.

       o  This profile follows the suggestion in the RTP spec that RTCP
          bandwidth may be specified separately from the session
          bandwidth and separately for active senders and passive

       o  RFC references are added for payload formats published after
          RFC 1890.

       o  A minimal TCP encapsulation is defined.

       o  The security considerations and full copyright sections were

       o  According to Peter Hoddie of Apple, only pre-1994 Macintosh
          used the 22254.54 rate and none the 11127.27 rate, so the
          latter was dropped from the discussion of suggested sampling

       o  Table 1 was corrected to move some values from the
          "ms/packet" column to the "default ms/packet" column where
          they belonged.

       o  A note has been added for G722 to clarify a discrepancy
          between the actual sampling rate and the RTP timestamp clock

       o  Small clarifications of the text have been made in several
          places, some in response to questions from readers. In

          - A definition for "media type" is given in Section 1.1 to
            allow the explanation of multiplexing RTP sessions in
            Section 6 to be more clear regarding the multiplexing of
            multiple media.

          - The explanation of how to determine the number of audio
            frames in a packet from the length was expanded.

          - More description of the allocation of bandwidth to SDES
            items is given.

          - The terms MUST, SHOULD, MAY, etc. are used as defined in RFC

       o  A second author for this document was added.

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10 Security Considerations

   Implementations using the profile defined in this specification are
   subject to the security considerations discussed in the RTP
   specification [1]. This profile does not specify any different
   security services other than giving rules for mapping characters in a
   user-provided pass phrase to canonical form.  The primary function of
   this profile is to list a set of data compression encodings for audio
   and video media.

   Confidentiality of the media streams is achieved by encryption.
   Because the data compression used with the payload formats described
   in this profile is applied end-to-end, encryption may be performed
   after compression so there is no conflict between the two operations.

   A potential denial-of-service threat exists for data encodings using
   compression techniques that have non-uniform receiver-end
   computational load. The attacker can inject pathological datagrams
   into the stream which are complex to decode and cause the receiver to
   be overloaded. However, the encodings described in this profile do
   not exhibit any significant non-uniformity.

   As with any IP-based protocol, in some circumstances a receiver may
   be overloaded simply by the receipt of too many packets, either
   desired or undesired. Network-layer authentication MAY be used to
   discard packets from undesired sources, but the processing cost of
   the authentication itself may be too high. In a multicast
   environment, pruning of specific sources may be implemented in future
   versions of IGMP [28] and in multicast routing protocols to allow a
   receiver to select which sources are allowed to reach it.

11 Full Copyright Statement

   Copyright (C) The Internet Society (1999). All Rights Reserved.

   This document and translations of it may be copied and furnished to
   others, and derivative works that comment on or otherwise explain it
   or assist in its implmentation may be prepared, copied, published and
   distributed, in whole or in part, without restriction of any kind,
   provided that the above copyright notice and this paragraph are
   included on all such copies and derivative works. However, this
   document itself may not be modified in any way, such as by removing
   the copyright notice or references to the Internet Society or other
   Internet organizations, except as needed for the purpose of
   developing Internet standards in which case the procedures for
   copyrights defined in the Internet Standards process must be
   followed, or as required to translate it into languages other than

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   The limited permissions granted above are perpetual and will not be
   revoked by the Internet Society or its successors or assigns.

   This document and the information contained herein is provided on an

12 Acknowledgements

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

13 Addresses of Authors

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

   Stephen L. Casner
   Cisco Systems, Inc.
   170 West Tasman Drive
   San Jose, CA 95134
   United States
   electronic mail: casner@cisco.com

A Bibliography

   [1] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: A
   transport protocol for real-time applications," Internet Draft,
   Internet Engineering Task Force, Feb. 1999 Work in progress, revision
   to RFC 1889.

   [2] S. Bradner, "Key words for use in RFCs to Indicate Requirement
   Levels,"  RFC 2119, Internet Engineering Task Force, Mar. 1997.

   [3] P. Hoschka, "MIME Type Registration of RTP Payload Types,"
   Internet Draft, Internet Engineering Task Force, Feb. 1999 Work in

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   [4] N. Freed, J. Klensin, and J. Postel, "Multipurpose Internet Mail
   Extensions (MIME) Part Four: Registration Procedures,"  RFC 2048,
   Internet Engineering Task Force, Nov. 1996.

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

   [6] Apple Computer, "Audio interchange file format AIFF-C," Aug.
   1991.  (also ftp://ftp.sgi.com/sgi/aiff-c.9.26.91.ps.Z).

   [7] 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.

   [8] 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.

   [9] 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.

   [10] 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.

   [11] 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.

   [12] 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.

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

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

   [15] S. M. Redl, M. K. Weber, and M. W. Oliphant, An Introduction to

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   GSM Boston: Artech House, 1995.

   [16] 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.

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

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

   [19] 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.

   [20] D. Tynan, "RTP payload format for BT.656 Video Encoding,"
   Request for Comments (Proposed Standard) RFC 2431, Internet
   Engineering Task Force, Oct. 1998.

   [21] 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.

   [22] L. Berc, W. Fenner, R. Frederick, and S. McCanne, "RTP payload
   format for JPEG-compressed video," Request for Comments (Proposed
   Standard) RFC 2435, Internet Engineering Task Force, Oct. 1996.

   [23] 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.

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

   [25] C. Bormann, L. Cline, G. Deisher, T. Gardos, C. Maciocco, D.
   Newell, J. Ott, G. Sullivan, S. Wenger, C. Zhu, "RTP Payload Format
   for the 1998 Version of ITU-T Rec. H.263 Video (H.263+)," Request for
   Comments (Proposed Standard) RFC 2429, Internet Engineering Task
   Force, Oct. 1998.

   [26] M. Civanlar, G. Cash, B. Haskell, "RTP Payload Format for
   Bundled MPEG," Request for Comments (Experimental) RFC 2343, Internet
   Engineering Task Force, May 1998.

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   [27] 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.

   [28] S. Deering, "Host Extensions for IP Multicasting," Request for
   Comments RFC 1112, STD 5, Internet Engineering Task Force, Aug. 1989.

   Current Locations of Related Resources

   Note: Several sections below refer to the ITU-T Software Tool Library
   (STL). It is available from the ITU Sales Service, Place des Nations,
   CH-1211 Geneve 20, Switzerland (also check http://www.itu.int. The
   ITU-T STL is covered by a license defined in ITU-T Recommendation
   G.191, "Software tools for speech and audio coding standardization".


   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


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


   The reference C code implementation defining the G.723.1 algorithm
   and its Annexes A, B, and C are available as an integral part of
   Recommendation G.723.1 from the ITU Sales Service, address listed
   above.  Both the algorithm and C code are covered by a specific
   license. The ITU-T Secretariat should be contacted to obtain such
   licensing information.


   G726-32 is specified in the ITU-T Recommendation G.726, "40, 32, 24,
   and 16 kb/s Adaptive Differential Pulse Code Modulation (ADPCM)". An
   implementation of the G.726 algorithm is available as part of the
   ITU-T STL, described above.


   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

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   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 [11].

                           Table of Contents

   1          Introduction ........................................    2
   1.1        Terminology .........................................    3
   2          RTP and RTCP Packet Forms and Protocol Behavior .....    3
   3          Registering Additional Encodings with IANA ..........    6
   4          Audio ...............................................    7
   4.1        Encoding-Independent Rules ..........................    7
   4.2        Operating Recommendations ...........................    9
   4.3        Guidelines for Sample-Based Audio Encodings .........    9
   4.4        Guidelines for Frame-Based Audio Encodings ..........   10
   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 ................................................   14
   4.5.6      G726-32 .............................................   14
   4.5.7      G728 ................................................   15
   4.5.8      G729 ................................................   16
   4.5.9      GSM .................................................   17    General Packaging Issues ............................   18    GSM variable names and numbers ......................   18
   4.5.10     GSM-HR ..............................................   18
   4.5.11     GSM-EFR .............................................   18
   4.5.12     L8 ..................................................   20
   4.5.13     L16 .................................................   20
   4.5.14     LPC .................................................   21
   4.5.15     MPA .................................................   21
   4.5.16     PCMA and PCMU .......................................   21

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   4.5.17     QCELP ...............................................   21
   4.5.18     RED .................................................   22
   4.5.19     VDVI ................................................   22
   5          Video ...............................................   23
   5.1        BT656 ...............................................   23
   5.2        CelB ................................................   23
   5.3        JPEG ................................................   24
   5.4        H261 ................................................   24
   5.5        H263 ................................................   24
   5.6        H263-1998 ...........................................   24
   5.7        MPV .................................................   24
   5.8        MP2T ................................................   24
   5.9        MP1S ................................................   25
   5.10       MP2P ................................................   25
   5.11       BMPEG ...............................................   25
   5.12       nv ..................................................   25
   6          Payload Type Definitions ............................   25
   7          RTP over TCP and Similar Byte Stream Protocols ......   26
   8          Port Assignment .....................................   26
   9          Changes from RFC 1890 ...............................   27
   10         Security Considerations .............................   30
   11         Full Copyright Statement ............................   30
   12         Acknowledgements ....................................   31
   13         Addresses of Authors ................................   31
   A          Bibliography ........................................   31

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