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rfc:rfc1890

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.

Abstract

 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.

Schulzrinne Standards Track [Page 1] RFC 1890 AV Profile January 1996

 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
 here.

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
 specification.
 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
      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.
 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.

Schulzrinne Standards Track [Page 2] RFC 1890 AV Profile January 1996

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

Schulzrinne Standards Track [Page 3] RFC 1890 AV Profile January 1996

 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
       companies);
      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
       below).
 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
       large.

Schulzrinne Standards Track [Page 4] RFC 1890 AV Profile January 1996

      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
 notation:
 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

Schulzrinne Standards Track [Page 5] RFC 1890 AV Profile January 1996

 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
 protocol.
 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
 receiver.

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.

Schulzrinne Standards Track [Page 6] RFC 1890 AV Profile January 1996

 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
 first).
 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.

Schulzrinne Standards Track [Page 7] RFC 1890 AV Profile January 1996

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
 [2,3,4,5].
 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:

Schulzrinne Standards Track [Page 8] RFC 1890 AV Profile January 1996

      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
       sample.
 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
 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 */
 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
 USA
 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".

Schulzrinne Standards Track [Page 9] RFC 1890 AV Profile January 1996

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
 France
 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
 USA
 electronic mail: don.hoffman@eng.sun.com

Schulzrinne Standards Track [Page 10] RFC 1890 AV Profile January 1996

 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

Schulzrinne Standards Track [Page 11] RFC 1890 AV Profile January 1996

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

Schulzrinne Standards Track [Page 12] RFC 1890 AV Profile January 1996

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

Schulzrinne Standards Track [Page 13] RFC 1890 AV Profile January 1996

 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.

Schulzrinne Standards Track [Page 14] RFC 1890 AV Profile January 1996

    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

Schulzrinne Standards Track [Page 15] RFC 1890 AV Profile January 1996

 system.

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
     1995.
[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.

Schulzrinne Standards Track [Page 16] 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
     1995.
[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
 acknowledged.

11. Author's Address

 Henning Schulzrinne
 GMD Fokus
 Hardenbergplatz 2
 D-10623 Berlin
 Germany
 EMail: schulzrinne@fokus.gmd.de

Schulzrinne Standards Track [Page 17] RFC 1890 AV Profile January 1996

 Current Locations of Related Resources
 UTF-8
 Information on the UCS Transformation Format 8 (UTF-8) is available
 at
          http://www.stonehand.com/unicode/standard/utf8.html
 1016
 An implementation is available at
            ftp://ftp.super.org/pub/speech/celp_3.2a.tar.Z
 DVI4
 An implementation is available from Jack Jansen at
              ftp://ftp.cwi.nl/local/pub/audio/adpcm.shar
 G721
 An implementation is available at
     ftp://gaia.cs.umass.edu/pub/hgschulz/ccitt/ccitt_tools.tar.Z
 GSM
 A reference implementation was written by Carsten Borman and Jutta
 Degener (TU Berlin, Germany). It is available at
          ftp://ftp.cs.tu-berlin.de/pub/local/kbs/tubmik/gsm/
 LPC
 An implementation is available at
          ftp://parcftp.xerox.com/pub/net-research/lpc.tar.Z

Schulzrinne Standards Track [Page 18]

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