|Title||RTP Profile for Audio and Video Conferences with Minimal Control
|Author||Audio-Video Transport Working Group, H. Schulzrinne
Network Working Group Audio-Video Transport Working Group
Request for Comments: 1890 H. Schulzrinne
Category: Standards Track GMD Fokus
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.
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.
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
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.
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.
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
o for proprietary encodings, contact information (postal and
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
o The encoding is of interest to the Internet community at
o It offers benefits compared to existing encodings and/or is
required for interoperation with existing, widely deployed
conferencing or multimedia systems.
o The description is sufficient to build a decoder.
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
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 , using the following
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 4.2 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.
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.
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.
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.
DVI4 is specified, with pseudo-code, in  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:
o The header contains the predicted value rather than the first
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
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.
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".
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
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
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.
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.
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.
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 ,
Section 3. The authors can be contacted at
Sun Microsystems, Inc.
2550 Garcia Avenue
Mountain View, California 94043-1100
electronic mail: email@example.com
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
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 .
A detailed description is given by Jayant and Noll .
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 .
PCMU is the encoding used for the Internet media type audio/basic. A
detailed description is given by Jayant and Noll .
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
The following video encodings are currently defined, with their
abbreviated names used for identification:
The CELL-B encoding is a proprietary encoding proposed by Sun
Microsystems. The byte stream format is described in work in
progress . The author can be contacted at
Michael F. Speer
Sun Microsystems Computer Corporation
2550 Garcia Ave MailStop UMPK14-305
Mountain View, CA 94043
electronic mail: firstname.lastname@example.org
The encoding is specified in ISO Standards 10918-1 and 10918-2. The
RTP payload format is as specified in work in progress . Further
information can be obtained from
Lawrence Berkeley National Laboratory
One Cyclotron Road
Berkeley, CA 94720
Phone: +1 510 486 7520
electronic mail: email@example.com
The encoding is specified in CCITT/ITU-T standard H.261. The
packetization and RTP-specific properties are described in work in
progress . Further information can be obtained from
Office NE 43-505
Telemedia, Networks and Systems
Laboratory for Computer Science
Massachusetts Institute of Technology
545 Technology Square
Cambridge, MA 02139
electronic mail: firstname.lastname@example.org
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 , Section 3. See the description of the MPA audio
encoding for contact information.
MP2T designates the use of MPEG-II transport streams, for either
audio or video. The encapsulation is described in work in progress,
, Section 2. See the description of the MPA audio encoding for
The encoding is implemented in the program 'nv', version 4, developed
at Xerox PARC by Ron Frederick. Further information is available from
Xerox Palo Alto Research Center
3333 Coyote Hill Road
Palo Alto, CA 94304
electronic mail: email@example.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
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 . 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
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
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
 Apple Computer, "Audio interchange file format AIFF-C," Aug.
1991. (also ftp://ftp.sgi.com/sgi/aiff-c.9.26.91.ps.Z).
 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-
 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.
 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.
 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.
 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.
 M. Mouly and M.-B. Pautet, The GSM system for mobile
communications Lassay-les-Chateaux, France: Europe Media
 J. Degener, "Digital speech compression," Dr. Dobb's Journal,
 S. M. Redl, M. K. Weber, and M. W. Oliphant, An Introduction to
GSM Boston: Artech House, 1995.
 D. Hoffman and V. Goyal, "RTP payload format for MPEG1/MPEG2
video," Work in Progress, Internet Engineering Task Force, June
 N. S. Jayant and P. Noll, Digital Coding of Waveforms--
Principles and Applications to Speech and Video Englewood Cliffs,
New Jersey: Prentice-Hall, 1984.
 M. F. Speer and D. Hoffman, "RTP payload format of CellB video
encoding," Work in Progress, Internet Engineering Task Force,
 W. Fenner, L. Berc, R. Frederick, and S. McCanne, "RTP
encapsulation of JPEG-compressed video," Work in Progress,
Internet Engineering Task Force, Mar. 1995.
 T. Turletti and C. Huitema, "RTP payload format for H.261 video
streams," Work in Progress, Internet Engineering Task Force, July
 H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: A
transport protocol for real-time applications." Work in Progress,
9. Security Considerations
Security issues are discussed in section 2.
The comments and careful review of Steve Casner are gratefully
11. Author's Address
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