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RFC1890 RTP Profile for Audio and Video Conferences with Minimal Control

RFC1890   RTP Profile for Audio and Video Conferences with Minimal Control    Audio-Video Transport Working Group, H. Schulzrinne [ January 1996 ] ( TXT = 37509 bytes)(Obsoleted by RFC3551)

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Network Working Group                Audio-Video Transport Working Group
Request for Comments: 1890                                H. Schulzrinne
Category: Standards Track                                      GMD Fokus
                                                            January 1996

    RTP Profile for Audio and Video Conferences with Minimal Control

Status of this Memo

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


   This memo describes a profile 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.

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

1.  Introduction

   This profile defines aspects of RTP left unspecified in the RTP
   Version 2 protocol definition (RFC 1889). 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.

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   Use of this profile occurs by use of the appropriate applications;
   there is no explicit indication by port number, protocol identifier
   or the like.

   Other profiles may make different choices for the items specified

2.  RTP and RTCP Packet Forms and Protocol Behavior

   The section "RTP Profiles and Payload Format Specification"
   enumerates a number of items that can be specified or modified in a
   profile. This section addresses these items. Generally, this profile
   follows the default and/or recommended aspects of the RTP

   RTP data header: The standard format of the fixed RTP data header is
        used (one marker bit).

   Payload types: Static payload types are defined in Section 6.

   RTP data header additions: No additional fixed fields are appended to
        the RTP data header.

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

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

   RTCP report interval: The suggested constants are to be used for the
        RTCP report interval calculation.

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

   SDES use: Applications may use any of the SDES items described.
        While CNAME information is sent every reporting interval, other
        items should be sent only every fifth reporting interval.

   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 key.)

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

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

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

   Underlying protocol: The profile specifies the use of RTP over
        unicast and multicast UDP. (This does not preclude the use of
        these definitions when RTP is carried by other lower-layer

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

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   Encapsulation: No encapsulation of RTP packets is specified.

3.  Registering Payload Types

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

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

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

        o any operating parameters or profiles;

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

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

        o the payload type value for this profile, if necessary (see

   Note that not all encodings to be used by RTP need to be assigned a
   static payload type. Non-RTP means beyond the scope of this memo
   (such as directory services or invitation protocols) may be used to
   establish a dynamic mapping between a payload type drawn from the
   range 96-127 and an encoding. For implementor convenience, this
   profile contains descriptions of encodings which do not currently
   have a static payload type assigned to them.

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

        o The encoding is of interest to the Internet community at

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

        o The description is sufficient to build a decoder.

4.  Audio

4.1 Encoding-Independent Recommendations

   For applications which send no packets during silence, the first
   packet of a talkspurt (first packet after a silence period) is
   distinguished by setting the marker bit in the RTP  data header.
   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 [1], using the following

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

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

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   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 4.2 and 4.3.

   The sampling frequency should be drawn from the set: 8000, 11025,
   16000, 22050, 24000, 32000, 44100 and 48000 Hz. (The Apple Macintosh
   computers have native sample rates of 22254.54 and 11127.27, which
   can be converted to 22050 and 11025 with acceptable quality by
   dropping 4 or 2 samples in a 20 ms frame.) However, most audio
   encodings are defined for a more restricted set of sampling
   frequencies. Receivers should be prepared to accept multi-channel
   audio, but may choose to only play a single channel.

   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, unless otherwise noted when describing the
   encoding. 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. This restriction allows reasonable buffer sizing for the

4.2 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.

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   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.

4.3 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 message. The receiver can tell the number of
   frames contained in a message since the frame duration is defined as
   part of the encoding.

   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.

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4.4 Audio Encodings

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

                 Table 1: Properties of Audio Encodings

   The characteristics of standard audio encodings are shown in Table 1
   and their payload types are listed in Table 2.

4.4.1 1016

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

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

4.4.2 DVI4

   DVI4 is specified, with pseudo-code, in [6] as the IMA ADPCM wave
   type. A specification titled "DVI ADPCM Wave Type" can also be found
   in the Microsoft Developer Network Development Library CD ROM
   published quarterly by Microsoft. The relevant section is found under
   Product Documentation, SDKs, Multimedia Standards Update, New
   Multimedia Data Types and Data Techniques, Revision 3.0, April 15,
   1994. However, the encoding defined here as DVI4 differs in two
   respects from these recommendations:

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

        o IMA ADPCM blocks contain 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

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

   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 */

   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

4.4.3 G721

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

4.4.4 G722

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

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

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4.4.6 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 [7,8,9]. 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

4.4.7 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.

4.4.8 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.

4.4.9 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.

4.4.10 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 work in progress [10],
   Section 3. The authors can be contacted at

   Don Hoffman
   Sun Microsystems, Inc.
   Mail-stop UMPK14-305
   2550 Garcia Avenue
   Mountain View, California 94043-1100
   electronic mail: don.hoffman@eng.sun.com

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

4.4.11 PCMA

   PCMA is specified in CCITT/ITU-T recommendation G.711. Audio data is
   encoded as eight bits per sample, after logarithmic scaling. Code to
   convert between linear and A-law companded data is available in [6].
   A detailed description is given by Jayant and Noll [11].

4.4.12 PCMU

   PCMU is specified in CCITT/ITU-T recommendation G.711. Audio data is
   encoded as eight bits per sample, after logarithmic scaling. Code to
   convert between linear and mu-law companded data is available in [6].
   PCMU is the encoding used for the Internet media type audio/basic.  A
   detailed description is given by Jayant and Noll [11].

4.4.13 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. 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

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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 work in
   progress [12].  The author can be contacted at

   Michael F. Speer
   Sun Microsystems Computer Corporation
   2550 Garcia Ave MailStop UMPK14-305
   Mountain View, CA 94043
   United States
   electronic mail: michael.speer@eng.sun.com

5.2 JPEG

The encoding is specified in ISO Standards 10918-1 and 10918-2. The
RTP payload format is as specified in work in progress [13].  Further
information can be obtained from

   Steven McCanne
   Lawrence Berkeley National Laboratory
   M/S 46A-1123
   One Cyclotron Road
   Berkeley, CA 94720
   United States
   Phone: +1 510 486 7520
   electronic mail: mccanne@ee.lbl.gov

5.3 H261

   The encoding is specified in CCITT/ITU-T standard H.261. The
   packetization and RTP-specific properties are described in work in
   progress [14]. Further information can be obtained from

   Thierry Turletti
   Office NE 43-505
   Telemedia, Networks and Systems
   Laboratory for Computer Science
   Massachusetts Institute of Technology
   545 Technology Square
   Cambridge, MA 02139
   United States
   electronic mail: turletti@clove.lcs.mit.edu

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5.4 MPV

   MPV designates the use MPEG-I and MPEG-II video encoding elementary
   streams as specified in ISO Standards ISO/IEC 11172 and 13818-2,
   respectively. The RTP payload format is as specified in work in
   progress [10], Section 3. See the description of the MPA audio
   encoding for contact information.

5.5 MP2T

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

5.6 nv

   The encoding is implemented in the program 'nv', version 4, developed
   at Xerox PARC by Ron Frederick. Further information is available from
   the author:

   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

   Table 2 defines this profile's static payload type values for the PT
   field of the RTP data header. A new RTP payload format specification
   may be registered with the IANA by name, and may also be assigned a
   static payload type value from the range marked in Section 3.

   In addition, payload type values in the range 96-127 may be defined
   dynamically through a conference control protocol, which is beyond
   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 time; the
   interleaving of several RTP payload types in a single RTP session is
   not allowed, but multiple RTP sessions may be used in parallel to
   send multiple media. The payload types currently defined in this

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   profile carry either audio or video, but not both. However, it is
   allowed to define payload types that combine several media, e.g.,
   audio and video, with appropriate separation in the payload format.
   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 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 [15]. The timestamp resolution must also be
   sufficient for the jitter estimate contained in the receiver reports.

   The standard video encodings and their payload types are listed in
   Table 2.

7.  Port Assignment

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

   Applications operating under this profile may use any such UDP port
   pair. For example, the port pair may be allocated randomly by a
   session management program. A single fixed port number pair cannot be
   required because multiple applications using this profile are likely
   to run on the same host, and there are some operating systems that do
   not allow multiple processes to use the same UDP port with different
   multicast addresses.

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      PT         encoding      audio/video    clock rate    channels
                 name          (A/V)          (Hz)          (audio)
      0          PCMU          A              8000          1
      1          1016          A              8000          1
      2          G721          A              8000          1
      3          GSM           A              8000          1
      4          unassigned    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         unassigned    A
      13         unassigned    A
      14         MPA           A              90000        (see text)
      15         G728          A              8000          1
      16--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--71     unassigned    ?
      72--76     reserved      N/A            N/A           N/A
      77--95     unassigned    ?
      96--127    dynamic       ?

   Table 2: Payload types (PT) for standard audio and video encodings

   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

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RFC 1890                       AV Profile                   January 1996


8. Bibliography

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

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

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

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

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

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

   [7] M. Mouly and M.-B. Pautet, The GSM system for mobile
       communications Lassay-les-Chateaux, France: Europe Media
       Duplication, 1993.

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

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

  [10] D. Hoffman and V. Goyal, "RTP payload format for MPEG1/MPEG2
       video," Work in Progress, Internet Engineering Task Force, June

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

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RFC 1890                       AV Profile                   January 1996

  [12] M. F. Speer and D. Hoffman, "RTP payload format of CellB video
       encoding," Work in Progress, Internet Engineering Task Force,
       Aug.  1995.

  [13] W. Fenner, L. Berc, R. Frederick, and S. McCanne, "RTP
       encapsulation of JPEG-compressed video," Work in Progress,
       Internet Engineering Task Force, Mar. 1995.

  [14] T. Turletti and C. Huitema, "RTP payload format for H.261 video
       streams," Work in Progress, Internet Engineering Task Force, July

  [15] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: A
       transport protocol for real-time applications." Work in Progress,
       Mar. 1995.

9.  Security Considerations

   Security issues are discussed in section 2.

10.  Acknowledgements

   The comments and careful review of Steve Casner are gratefully

11.  Author's Address

   Henning Schulzrinne
   GMD Fokus
   Hardenbergplatz 2
   D-10623 Berlin

   EMail: schulzrinne@fokus.gmd.de

Schulzrinne                 Standards Track                    [Page 17]
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   Current Locations of Related Resources


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



   An implementation is available at



   An implementation is available from Jack Jansen at



   An implementation is available at



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



   An implementation is available at


Schulzrinne                 Standards Track                    [Page 18]



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