Internet Engineering Task Force AVT WG
Internet Draft Schulzrinne
ietf-avt-profile-new-05.txt Columbia U.
February 26, 1999
Expires: August 26, 1999
RTP Profile for Audio and Video Conferences with Minimal Control
STATUS OF THIS MEMO
This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026.
Internet-Drafts are working documents of the Internet Engineering
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ABSTRACT
This memorandum is a revision of RFC 1890 in preparation
for advancement from Proposed Standard to Draft Standard
status. Readers are encouraged to use the PostScript form
of this draft to see where changes from RFC 1890 are
marked by change bars.
This document describes a profile called "RTP/AVP" for
the use of the real-time transport protocol (RTP),
version 2, and the associated control protocol, RTCP,
within audio and video multiparticipant conferences with
minimal control. It provides interpretations of generic
fields within the RTP specification suitable for audio
and video conferences. In particular, this document
defines a set of default mappings from payload type
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numbers to encodings.
This document also describes how audio and video data may
be carried within RTP. It defines a set of standard
encodings and their names when used within RTP. The
descriptions provide pointers to reference
implementations and the detailed standards. This document
is meant as an aid for implementors of audio, video and
other real-time multimedia applications.
Resolution of Open Issues
[Note to the RFC Editor: This section is to be deleted when this
draft is published as an RFC but is shown here for reference during
the Last Call. All RFC XXXX should be filled in with the number of
the RTP specification RFC submitted for Draft Standard status, and
all RFC YYYY should be filled in with the number of the draft
specifying MIME registration of RTP payload types as it is submitted
for Proposed Standard status.]
Readers are directed to Appendix 9, Changes from RFC 1890, for a
listing of the changes that have been made in this draft. The
changes from RFC 1890 are marked with change bars in the PostScript
form of this draft.
The revisions in this draft are intended to be complete for Working
Group last call. The following open issues from previous drafts have
been addressed:
o The procedure for registering encoding names as MIME subtypes
is outlined here and referenced in a separate RFC-to-be that
may also serve to specify how (some of) the encodings here may
be used with mail and other not-RTP transports.
o This profile follows the suggestion in the RTP spec that RTCP
bandwidth may be specified separately from the session
bandwidth and separately for active senders and passive
receivers.
o No specific action is taken in this document to address
generic payload formats; it is assumed that if any generic
payload formats are developed, they can be specified in
separate RFCs and that the session parameters they require for
operation can be specified in the MIME registration of those
formats.
1 Introduction
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This profile defines aspects of RTP left unspecified in the RTP
Version 2 protocol definition (RFC XXXX) [1]. This profile is
intended for the use within audio and video conferences with minimal
session control. In particular, no support for the negotiation of
parameters or membership control is provided. The profile is expected
to be useful in sessions where no negotiation or membership control
are used (e.g., using the static payload types and the membership
indications provided by RTCP), but this profile may also be useful in
conjunction with a higher-level control protocol.
Use of this profile may be implicit in the use of the appropriate
applications; there may be no explicit indication by port number,
protocol identifier or the like. Applications such as session
directories should refer to this profile as "RTP/AVP".
Other profiles may make different choices for the items specified
here.
This document also defines a set of encodings and payload formats for
audio and video.
1.1 Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [2] and
indicate requirement levels for implementations compliant with this
RTP profile.
This draft defines the term media type as dividing encodings of audio
and video content into three classes: audio, video and audio/video
(interleaved).
2 RTP and RTCP Packet Forms and Protocol Behavior
The section "RTP Profiles and Payload Format Specification" of RFC
XXXX enumerates a number of items that can be specified or modified
in a profile. This section addresses these items. Generally, this
profile follows the default and/or recommended aspects of the RTP
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.
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RTP data header extensions: No RTP header extensions are defined, but
applications operating under this profile MAY use such
extensions. Thus, applications SHOULD NOT assume that the RTP
header X bit is always zero and SHOULD be prepared to ignore the
header extension. If a header extension is defined in the
future, that definition MUST specify the contents of the first
16 bits in such a way that multiple different extensions can be
identified.
RTCP packet types: No additional RTCP packet types are defined by
this profile specification.
RTCP report interval: The suggested constants are to be used for the
RTCP report interval calculation. Sessions operating under this
profile MAY specify a separate parameter for the RTCP traffic
bandwidth rather than using the default fraction of the session
bandwidth. The RTCP traffic bandwidth may be divided into two
separate session parameters for those participants which are
active data senders and those which are not. Following the
recommendation in the RTP specification [1] that 1/4 of the RTCP
bandwidth be dedicated to data senders, the RECOMMENDED default
values for these two parameters would be 1.25% and 3.75%,
respectively. For a particular session, the RTCP bandwidth for
non-data-senders MAY be set to zero when operating on
unidirectional links or for sessions that don't require feedback
on the quality of reception. The RTCP bandwidth for data senders
SHOULD be kept non-zero so that sender reports can still be sent
for inter-media synchronization and to identify the source by
CNAME. The means by which the one or two session parameters for
RTCP bandwidth are specified is beyond the scope of this memo.
SR/RR extension: No extension section is defined for the RTCP SR or
RR packet.
SDES use: Applications MAY use any of the SDES items described in the
RTP specification. While CNAME information MUST be sent every
reporting interval, other items SHOULD only be sent every third
reporting interval, with NAME sent seven out of eight times
within that slot and the remaining SDES items cyclically taking
up the eighth slot, as defined in Section 6.2.2 of the RTP
specification. In other words, NAME is sent in RTCP packets 1,
4, 7, 10, 13, 16, 19, while, say, EMAIL is used in RTCP packet
22.
Security: The RTP default security services are also the default
under this profile.
String-to-key mapping: A user-provided string ("pass phrase") is
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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 RECOMMENDED that pass phrases be restricted to ASCII letters,
digits, the hyphen, and white space to reduce the the chance of
transcription errors when conveying keys by phone, fax, telex or
email.
The pass phrase MAY be preceded by a specification of the encryption
algorithm. Any characters up to the first slash (ASCII 0x2f) are
taken as the name of the encryption algorithm. The encryption format
specifiers SHOULD be drawn from RFC 1423 or any additional
identifiers registered with IANA. If no slash is present, DES-CBC is
assumed as default. The encryption algorithm specifier is case
sensitive.
The pass phrase typed by the user is transformed to a canonical form
before applying the hash algorithm. For that purpose, we define
`white space' to be the ASCII space, formfeed, newline, carriage
return, tab, or vertical tab as well as all characters contained in
the Unicode space characters table. The transformation consists of
the following steps: (1) convert the input string to the ISO 10646
character set, using the UTF-8 encoding as specified in Annex P to
ISO/IEC 10646-1:1993 (ASCII characters require no mapping, but ISO
8859-1 characters do); (2) remove leading and trailing white space
characters; (3) replace one or more contiguous white space characters
by a single space (ASCII or UTF-8 0x20); (4) convert all letters to
lower case and replace sequences of characters and non-spacing
accents with a single character, where possible. A minimum length of
16 key characters (after applying the transformation) SHOULD be
enforced by the application, while applications MUST allow up to 256
characters of input.
Underlying protocol: The profile specifies the use of RTP over
unicast and multicast UDP as well as TCP. (This does not
preclude the use of these definitions when RTP is carried by
other lower-layer protocols.)
Transport mapping: The standard mapping of RTP and RTCP to
transport-level addresses is used.
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Encapsulation: A minimal TCP encapsulation is defined.
3 Registering Additional Encodings with IANA
This profile defines a set of encodings and assigns names to them.
These encodings are comprised of a particular media data compression
or representation plus an payload format for encapsulation within
RTP. It is expected that additional encodings beyond the set defined
here will be created in the future. These additional encodings may be
registered with the Internet Assigned Numbers Authority (IANA) as
MIME subtype names under the "audio" and "video" MIME types through
the MIME registration procedure as specified in RFC 2048 [3].
The MIME registration procedure was originally designed for transport
of multimedia information via asynchronous Internet mail, but the
MIME namespace now provides identification for other transport modes
as well. Some additional parameters are required in the registration
procedure to specify how a particular media type is transported over
RTP. These extensions to the registration procedure, along with
registrations for the encodings defined in this memo, are given in
RFC YYYY [4].
The encoding definitions contained in this memo provide examples of
the information needed:
o the name of the encoding; the names defined here are 3 or 4
characters long to allow a compact representation if needed;
o a description of encoding, including in particular the RTP
timestamp clock rate (or multiple rates for audio encodings
with multiple sampling rates), as may be provided by a separte
RTP payload format specification;
o any additional or optional parameters, such as the number of
channels for an audio encoding;
o a reference to a further description of the data compression
format itself, if available;
o contact information and an indication of who has change
control over the encoding (this is part of all MIME
registrations).
In addition to assigning names to encodings, this profile also also
assigns static RTP payload types to some of them. However, the
payload type number space is relatively small and cannot accommodate
assignments for all existing and future encodings. During the early
stages of RTP development, it was necessary to use statically
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assigned payload types because no other mechanism had been specified
to bind encodings to payload types. It was anticipated that non-RTP
means beyond the scope of this memo (such as directory services or
invitation protocols) would be specified to establish a dynamic
mapping between a payload type and an encoding. Now, mechanisms for
defining dynamic payload type bindings have been specified in the
Session Description Protocol (SDP), RFC 2327 [5], and in other
protocols such as ITU-T recommendation H.323/H.245. These mechanisms
associate the registered name of the encoding/payload format, along
with any additional required parameters such as the RTP timestamp
clock rate and number of channels, to a payload type number. This
association is effective only for the duration of the RTP session in
which the dynamic payload type binding is made. This association
applies only to the RTP session for which it is made, thus the
numbers can be re-used for different encodings in different sessions
so the number space limitation is avoided.
This profile reserves payload type numbers in the range 96-127
exclusively for dynamic assignment. Applications should first use
values in this range for dynamic payload types. Only applications
which need to define more than 32 dynamic payload types may bind
codes below 96, in which case it is RECOMMENDED that unassigned
payload type numbers be used first. However, the statically assigned
payload types are default bindings and may be dynamically bound to
new encodings if needed. Redefining payload types below 96 may cause
incorrect operation if an attempt is made to join a session without
obtaining session description information that defines the dynamic
payload types.
Dynamic payload types SHOULD NOT be used without a well-defined
mechanism to indicate the mapping. Systems that expect to
interoperate with others operating under this profile SHOULD NOT make
their own assignments of proprietary encodings to particular, fixed
payload types.
This specification establishes the policy that no additional static
payload types will be assigned beyond the ones defined in this
document. Establishing this policy avoids the problem of trying to
create a set of criteria for accepting static assignments and
encourages the implementation and deployment of the dynamic payload
type mechanisms.
4 Audio
4.1 Encoding-Independent Rules
For applications which send either no packets or comfort-noise
packets during silence, the first packet of a talkspurt, that is, the
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first packet after a silence period, SHOULD be distinguished by
setting the marker bit in the RTP data header to one. The marker bits
in all other packets is zero. The beginning of a talkspurt MAY be
used to adjust the playout delay to reflect changing network delays.
Applications without silence suppression MUST set the marker bit to
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 [6], 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
Samples for all channels belonging to a single sampling instant MUST
be within the same packet. The interleaving of samples from different
channels depends on the encoding. General guidelines are given in
Section 4.3 and 4.4.
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The sampling frequency SHOULD be drawn from the set: 8000, 11025,
16000, 22050, 24000, 32000, 44100 and 48000 Hz. (Older Apple
Macintosh computers had a native sample rate of 22254.54 Hz, which
can be converted to 22050 with acceptable quality by dropping 4
samples in a 20 ms frame.) However, most audio encodings are defined
for a more restricted set of sampling frequencies. Receivers SHOULD
be prepared to accept multi-channel audio, but MAY choose to only
play a single channel.
4.2 Operating Recommendations
The following recommendations are default operating parameters.
Applications SHOULD be prepared to handle other values. The ranges
given are meant to give guidance to application writers, allowing a
set of applications conforming to these guidelines to interoperate
without additional negotiation. These guidelines are not intended to
restrict operating parameters for applications that can negotiate a
set of interoperable parameters, e.g., through a conference control
protocol.
For packetized audio, the default packetization interval SHOULD have
a duration of 20 ms or one frame, whichever is longer, unless
otherwise noted in Table 1 (column "ms/packet"). The packetization
interval determines the minimum end-to-end delay; longer packets
introduce less header overhead but higher delay and make packet loss
more noticeable. For non-interactive applications such as lectures or
for links with severe bandwidth constraints, a higher packetization
delay MAY be used. A receiver SHOULD accept packets representing
between 0 and 200 ms of audio data. (For framed audio encodings, a
receiver SHOULD accept packets with 200 ms divided by the frame
duration, rounded up.) This restriction allows reasonable buffer
sizing for the receiver.
4.3 Guidelines for Sample-Based Audio Encodings
In sample-based encodings, each audio sample is represented by a
fixed number of bits. Within the compressed audio data, codes for
individual samples may span octet boundaries. An RTP audio packet may
contain any number of audio samples, subject to the constraint that
the number of bits per sample times the number of samples per packet
yields an integral octet count. Fractional encodings produce less
than one octet per sample.
The duration of an audio packet is determined by the number of
samples in the packet.
For sample-based encodings producing one or more octets per sample,
samples from different channels sampled at the same sampling instant
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SHOULD be packed in consecutive octets. For example, for a two-
channel encoding, the octet sequence is (left channel, first sample),
(right channel, first sample), (left channel, second sample), (right
channel, second sample), .... For multi-octet encodings, octets
SHOULD be transmitted in network byte order (i.e., most significant
octet first).
The packing of sample-based encodings producing less than one octet
per sample is encoding-specific.
The RTP timestamp reflects the instant at which the first sample in
the packet was sampled, that is, the oldest information in the
packet.
4.4 Guidelines for Frame-Based Audio Encodings
Frame-based encodings encode a fixed-length block of audio into
another block of compressed data, typically also of fixed length. For
frame-based encodings, the sender MAY choose to combine several such
frames into a single RTP packet. The receiver can tell the number of
frames contained in an RTP packet, if all the frames have the same
length, by dividing the RTP payload length by the audio frame size
which is defined as part of the encoding. This does not work when
carrying frames of different sizes unless the frame sizes are
relatively prime. If not, the frames MUST indicate their size.
For frame-based codecs, the channel order is defined for the whole
block. That is, for two-channel audio, right and left samples SHOULD
be coded independently, with the encoded frame for the left channel
preceding that for the right channel.
All frame-oriented audio codecs SHOULD be able to encode and decode
several consecutive frames within a single packet. Since the frame
size for the frame-oriented codecs is given, there is no need to use
a separate designation for the same encoding, but with different
number of frames per packet.
RTP packets SHALL contain a whole number of frames, with frames
inserted according to age within a packet, so that the oldest frame
(to be played first) occurs immediately after the RTP packet header.
The RTP timestamp reflects the instant at which the first sample in
the first frame was sampled, that is, the oldest information in the
packet.
4.5 Audio Encodings
The characteristics of the audio encodings described in this document
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name of sampling default
encoding sample/frame bits/sample rate ms/frame ms/packet
____________________________________________________________________________
1016 frame N/A 8,000 30 30
CN frame N/A var.
DVI4 sample 4 var. 20
G722 sample 8 16,000 20
G723 frame N/A 8,000 30 30
G726-16 sample 2 8,000 20
G726-24 sample 3 8,000 20
G726-32 sample 4 8,000 20
G726-40 sample 5 8,000 20
G727-16 sample 2 8,000 20
G727-24 sample 3 8,000 20
G727-32 sample 4 8,000 20
G727-40 sample 5 8,000 20
G728 frame N/A 8,000 2.5 20
G729 frame N/A 8,000 10 20
GSM frame N/A 8,000 20 20
GSM-HR frame N/A 8,000 20 20
GSM-EFR frame N/A 8,000 20 20
L8 sample 8 var. 20
L16 sample 16 var. 20
LPC frame N/A 8,000 20 20
MPA frame N/A var. 20
PCMA sample 8 var. 20
PCMU sample 8 var. 20
QCELP frame N/A 8,000 20
SX7300P frame N/A 8,000 15 30
SX8300P frame N/A 8,000 15 30
SX9600P frame N/A 8,000 15 30
VDVI sample var. var. 20
Table 1: Properties of Audio Encodings (N/A: not applicable; var.:
variable)
are shown in Table 1; they are listed in order of their payload type
in Table 4. While most audio codecs are only specified for a fixed
sampling rate, some sample-based algorithms (indicated by an entry of
"var." in the sampling rate column of Table 1) may be used with
different sampling rates, resulting in different coded bit rates.
When used with a sampling rate other than that for which a static
payload type is defined, non-RTP means beyond the scope of this memo
MUST be used to define a dynamic payload type and MUST indicate the
selected sampling rate.
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4.5.1 1016
Encoding 1016 is a frame based encoding using code-excited linear
prediction (CELP) and is specified in Federal Standard FED-STD 1016
[7,8,9,10].
4.5.2 CN
The CN (comfort noise) packet contains a single-octet message to the
receiver to play comfort noise at the absolute level specified. This
message would normally be sent once at the beginning of a silence
period (which also indicates the transition from speech to silence),
but the rate of noise level updates is implementation specific. The
magnitude of the noise level is packed into the least significant
bits of the noise-level payload, as shown below.
The noise level is expressed in -dBov, with values from 0 to 127
representing 0 to -127 dBov. dBov is the level relative to the
overload of the system. (Note: Representation relative to the
overload point of a system is particularly useful for digital
implementations, since one does not need to know the relative
calibration of the analog circuitry.) For example, in a 16-bit linear
PCM system (L16), a signal with 0 dBov represents a square wave with
the maximum possible amplitude (+/-32767), and -63 dBov corresponds
to -58 dBm0 in a standard telephone system. (dBm is the power level
in decibels relative to 1 mW, with an impedance of 600 Ohms.)
0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|0| level |
+-+-+-+-+-+-+-+-+
The RTP header for the comfort noise packet SHOULD be constructed as
if the comfort noise were an independent codec. Thus, the RTP
timestamp designates the beginning of the silence period. A static
payload type is assigned for a sampling rate of 8,000 Hz; if other
sampling rates are needed, they MUST be defined through dynamic
payload types. The RTP packet SHOULD NOT have the marker bit set.
The CN payload type is primarily for use with L16, DVI4, PCMA, PCMU
and other audio codecs that do not support comfort noise as part of
the codec itself. G.723.1 and G.729 have their own comfort noise
systems as part of Annexes A (G.723.1) and B (G.729), respectively.
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4.5.3 DVI4
DVI4 is specified, with pseudo-code, in [11] as the IMA ADPCM wave
type.
However, the encoding defined here as DVI4 differs in three respects
from this recommendation:
o The RTP DVI4 header contains the predicted value rather than
the first sample value contained the IMA ADPCM block header.
o IMA ADPCM blocks contain an odd number of samples, since the
first sample of a block is contained just in the header
(uncompressed), followed by an even number of compressed
samples. DVI4 has an even number of compressed samples only,
using the `predict' word from the header to decode the first
sample.
o For DVI4, the 4-bit samples are packed with the first sample
in the four most significant bits and the second sample in the
four least significant bits. In the IMA ADPCM codec, the
samples are packed in the opposite order.
Each packet contains a single DVI block. This profile only defines
the 4-bit-per-sample version, while IMA also specifies a 3-bit-per-
sample encoding.
The "header" word for each channel has the following structure:
int16 predict; /* predicted value of first sample
from the previous block (L16 format) */
u_int8 index; /* current index into stepsize table */
u_int8 reserved; /* set to zero by sender, ignored by receiver */
Each octet following the header contains two 4-bit samples, thus the
number of samples per packet MUST be even because there is no means
to indicate a partially filled last octet.
Packing of samples for multiple channels is for further study.
The document IMA Recommended Practices for Enhancing Digital Audio
Compatibility in Multimedia Systems (version 3.0) contains the
algorithm description. It is available from
Interactive Multimedia Association
48 Maryland Avenue, Suite 202
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Annapolis, MD 21401-8011
USA
phone: +1 410 626-1380
4.5.4 G722
G722 is specified in ITU-T Recommendation G.722, "7 kHz audio-coding
within 64 kbit/s".
4.5.5 G723
G723 is specified in ITU Recommendation G.723.1, "Dual-rate speech
coder for multimedia communications transmitting at 5.3 and 6.3
kbit/s". The G.723.1 5.3/6.3 kbit/s codec was defined by the ITU-T as
a mandatory codec for ITU-T H.324 GSTN videophone terminal
applications. The algorithm has a floating point specification in
Annex B to G.723.1, a silence compression algorithm in Annex A to
G.723.1 and an encoded signal bit-error sensitivity specification in
G.723.1 Annex C.
This Recommendation specifies a coded representation that can be used
for compressing the speech signal component of multi-media services
at a very low bit rate. Audio is encoded in 30 ms frames, with an
additional delay of 7.5 ms due to look-ahead. A G.723.1 frame can be
one of three sizes: 24 octets (6.3 kb/s frame), 20 octets (5.3 kb/s
frame), or 4 octets. These 4-octet frames are called SID frames
(Silence Insertion Descriptor) and are used to specify comfort noise
parameters. There is no restriction on how 4, 20, and 24 octet frames
are intermixed. The least significant two bits of the first octet in
the frame determine the frame size and codec type:
bits content octets/frame
00 high-rate speech (6.3 kb/s) 24
01 low-rate speech (5.3 kb/s) 20
10 SID frame 4
11 reserved
It is possible to switch between the two rates at any 30 ms frame
boundary. Both (5.3 kb/s and 6.3 kb/s) rates are a mandatory part of
the encoder and decoder. This coder was optimized to represent speech
with near-toll quality at the above rates using a limited amount of
complexity.
The packing of the encoded bit stream into octets and the
transmission order of the octets is specified in G.723.1.
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4.5.6 G726-16, G726-24, G726-32, G726-40
ITU-T Recommendation G.726 describes, among others, the algorithm
recommended for conversion of a single 64 kbit/s A-law or mu-law PCM
channel encoded at 8000 samples/sec to and from a 32 kbit/s channel.
The conversion is applied to the PCM stream using an Adaptive
Differential Pulse Code Modulation (ADPCM) transcoding technique.
G.726 describes codecs operating at 16 kb/s (2 bits/sample), 24 kb/s
(3 bits/sample), 32 kb/s (4 bits/sample), 40 kb/s (5 bits/sample).
These encodings are labeled G726-16, G726-24, G726-32 and G726-40,
respectively.
Note: In 1990, ITU-T Recommendation G.721 was merged with
Recommendation G.723 into ITU-T Recommendation G.726. Thus, G726-32
designates the same algorithm as G721 in RFC 1890.
No payload-specific header information SHALL be included as part of
the audio data. The 4-bit code words of the G726-32 encoding MUST be
packed into octets as follows: the first code word is placed in the
four least significant bits of the first octet, with the least
significant bit of the code word in the least significant bit of the
octet; the second code word is placed in the four most significant
bits of the first octet, with the most significant bit of the code
word in the most significant bit of the octet. Subsequent pairs of
the code words SHALL be packed in the same way into successive
octets, with the first code word of each pair placed in the least
significant four bits of the octet. The number of samples per packet
MUST be even because there is no means to indicate a partially filled
last octet.
4.5.7 G727-16, G727-24, G727-32, G727-40
ITU-T Recommendation G.727, "5-, 4-, 3- and 2-bits sample embedded
adaptive differential pulse code modulation (ADPCM)", specifies an
embedded ADPCM algorithm which has the intrinsic capability of
dropping bits in the encoded words to alleviate network congestion
conditions. The algorithm, although not bitstream compatible with
G.726, was based and has a structure similar to the G.726 ADPCM
algorithm.
4.5.8 G728
G728 is specified in ITU-T Recommendation G.728, "Coding of speech at
16 kbit/s using low-delay code excited linear prediction".
A G.278 encoder translates 5 consecutive audio samples into a 10-bit
codebook index, resulting in a bit rate of 16 kb/s for audio sampled
at 8,000 samples per second. The group of five consecutive samples is
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called a vector. Four consecutive vectors, labeled V1 to V4 (where V1
is to be played first by the receiver), build one G.728 frame. The
four vectors of 40 bits are packed into 5 octets, labeled B1 through
B5. B1 SHALL be placed first in the RTP packet.
Referring to the figure below, the principle for bit order is
"maintenance of bit significance". Bits from an older vector are more
significant than bits from newer vectors. The MSB of the frame goes
to the MSB of B1 and the LSB of the frame goes to LSB of B5.
1 2 3 3
0 0 0 0 9
++++++++++++++++++++++++++++++++++++++++
<---V1---><---V2---><---V3---><---V4---> vectors
<--B1--><--B2--><--B3--><--B4--><--B5--> octets
<------------- frame 1 ---------------->
In particular, B1 contains the eight most significant bits of V1,
with the MSB of V1 being the MSB of B1. B2 contains the two least
significant bits of V1, the more significant of the two in its MSB,
and the six most significant bits of V2. B1 SHALL be placed first in
the RTP packet and B5 last.
4.5.9 G729
G729 is specified in ITU-T Recommendation G.729, "Coding of speech at
8 kbit/s using conjugate structure-algebraic code excited linear
prediction (CS-ACELP)". A reduced-complexity version of the G.729
algorithm is specified in Annex A to Rec. G.729. The speech coding
algorithms in the main body of G.729 and in G.729 Annex A are fully
interoperable with each other, so there is no need to further
distinguish between them. The G.729 and G.729 Annex A codecs were
optimized to represent speech with high quality, where G.729 Annex A
trades some speech quality for an approximate 50% complexity
reduction [12].
A voice activity detector (VAD) and comfort noise generator (CNG)
algorithm in Annex B of G.729 is recommended for digital simultaneous
voice and data applications and can be used in conjunction with G.729
or G.729 Annex A. A G.729 or G.729 Annex A frame contains 10 octets,
while the G.729 Annex B comfort noise frame occupies 2 octets:
0 1
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
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+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|L| LSF1 | LSF2 | GAIN |R|
|S| | | |E|
|F|0 1 2 3 4|0 1 2 3|0 1 2 3 4|S|
|0| | | |V| RESV = Reserved (zero)
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
An RTP packet may consist of zero or more G.729 or G.729 Annex A
frames, followed by zero or one G.729 Annex B payloads. The presence
of a comfort noise frame can be deduced from the length of the RTP
payload.
The transmitted parameters of a G.729/G.729A 10-ms frame, consisting
of 80 bits, are defined in Recommendation G.729, Table 8/G.729.
The mapping of the these parameters is given below. Bits are numbered
as Internet order, that is, the most significant bit is bit 0.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|L| L1 | L2 | L3 | P1 |P| C1 |
|0| | | | |0| |
| |0 1 2 3 4 5 6|0 1 2 3 4|0 1 2 3 4|0 1 2 3 4 5 6 7| |0 1 2 3 4|
| | | | | | | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
4 5 6
2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| C1 | S1 | GA1 | GB1 | P2 | C2 |
| | | | | | |
|5 6 7 8 9 1 1 1|0 1 2 3|0 1 2|0 1 2 3|0 1 2 3 4|0 1 2 3 4 5 6 7|
| 0 1 2| | | | | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
7
4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| C2 | S2 | GA2 | GB2 |
| | | | |
|8 9 1 1 1|0 1 2 3|0 1 2|0 1 2 3|
| 0 1 2| | | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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4.5.10 GSM
GSM (group speciale mobile) denotes the European GSM 06.10 standard
for full-rate speech transcoding, ETS 300 961, which is based on
RPE/LTP (residual pulse excitation/long term prediction) coding at a
rate of 13 kb/s [13,14,15]. The text of the standard can be obtained
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
Blocks of 160 audio samples are compressed into 33 octets, for an
effective data rate of 13,200 b/s.
4.5.10.1 General Packaging Issues
The GSM standard specifies the bit stream produced by the codec, but
does not specify how these bits should be packed for transmission.
Some software implementations of the GSM codec use a different
packing than that specified here.
In the GSM packing used by RTP, the bits SHALL be packed beginning
from the most significant bit. Every 160 sample GSM frame is coded
into one 33 octet (264 bit) buffer. Every such buffer begins with a 4
bit signature (0xD), followed by the MSB encoding of the fields of
the frame. The first octet thus contains 1101 in the 4 most
significant bits (0-3) and the 4 most significant bits of F1 (0-3) in
the 4 least significant bits (4-7). The second octet contains the 2
least significant bits of F1 in bits 0-1, and F2 in bits 2-7, and so
on. The order of the fields in the frame is described in Table 2.
4.5.10.2 GSM variable names and numbers
In the RTP encoding we have the bit pattern described in Table 3,
where F.i signifies the ith bit of the field F, bit 0 is the most
significant bit, and the bits of every octet are numbered from 0 to 7
from most to least significant.
4.5.11 GSM-HR
GSM-HR denotes GSM 06.20 half rate speech transcoding, specified in
ETS 300 969 which is available from ETSI at the address given in
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field field name bits field field name bits
__________________________________________________________
1 LARc[0] 6 39 xmc[22] 3
2 LARc[1] 6 40 xmc[23] 3
3 LARc[2] 5 41 xmc[24] 3
4 LARc[3] 5 42 xmc[25] 3
5 LARc[4] 4 43 Nc[2] 7
6 LARc[5] 4 44 bc[2] 2
7 LARc[6] 3 45 Mc[2] 2
8 LARc[7] 3 46 xmaxc[2] 6
9 Nc[0] 7 47 xmc[26] 3
10 bc[0] 2 48 xmc[27] 3
11 Mc[0] 2 49 xmc[28] 3
12 xmaxc[0] 6 50 xmc[29] 3
13 xmc[0] 3 51 xmc[30] 3
14 xmc[1] 3 52 xmc[31] 3
15 xmc[2] 3 53 xmc[32] 3
16 xmc[3] 3 54 xmc[33] 3
17 xmc[4] 3 55 xmc[34] 3
18 xmc[5] 3 56 xmc[35] 3
19 xmc[6] 3 57 xmc[36] 3
20 xmc[7] 3 58 xmc[37] 3
21 xmc[8] 3 59 xmc[38] 3
22 xmc[9] 3 60 Nc[3] 7
23 xmc[10] 3 61 bc[3] 2
24 xmc[11] 3 62 Mc[3] 2
25 xmc[12] 3 63 xmaxc[3] 6
26 Nc[1] 7 64 xmc[39] 3
27 bc[1] 2 65 xmc[40] 3
28 Mc[1] 2 66 xmc[41] 3
29 xmaxc[1] 6 67 xmc[42] 3
30 xmc[13] 3 68 xmc[43] 3
31 xmc[14] 3 69 xmc[44] 3
32 xmc[15] 3 70 xmc[45] 3
33 xmc[16] 3 71 xmc[46] 3
34 xmc[17] 3 72 xmc[47] 3
35 xmc[18] 3 73 xmc[48] 3
36 xmc[19] 3 74 xmc[49] 3
37 xmc[20] 3 75 xmc[50] 3
38 xmc[21] 3 76 xmc[51] 3
Table 2: Ordering of GSM variables
Section 4.5.10. This codec has a frame length of 112 bits (14
octets). Packing of the fields in the codec bit stream into octets
for transmission in RTP is done in a manner similar to that specified
here for the original GSM 06.10 codec and is specified in ETSI
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Octet Bit 0 Bit 1 Bit 2 Bit 3 Bit 4 Bit 5 Bit 6 Bit 7
_____________________________________________________________________________________________
0 1 1 0 1 LARc0.0 LARc0.1 LARc0.2 LARc0.3
1 LARc0.4 LARc0.5 LARc1.0 LARc1.1 LARc1.2 LARc1.3 LARc1.4 LARc1.5
2 LARc2.0 LARc2.1 LARc2.2 LARc2.3 LARc2.4 LARc3.0 LARc3.1 LARc3.2
3 LARc3.3 LARc3.4 LARc4.0 LARc4.1 LARc4.2 LARc4.3 LARc5.0 LARc5.1
4 LARc5.2 LARc5.3 LARc6.0 LARc6.1 LARc6.2 LARc7.0 LARc7.1 LARc7.2
5 Nc0.0 Nc0.1 Nc0.2 Nc0.3 Nc0.4 Nc0.5 Nc0.6 bc0.0
6 bc0.1 Mc0.0 Mc0.1 xmaxc00 xmaxc01 xmaxc02 xmaxc03 xmaxc04
7 xmaxc05 xmc0.0 xmc0.1 xmc0.2 xmc1.0 xmc1.1 xmc1.2 xmc2.0
8 xmc2.1 xmc2.2 xmc3.0 xmc3.1 xmc3.2 xmc4.0 xmc4.1 xmc4.2
9 xmc5.0 xmc5.1 xmc5.2 xmc6.0 xmc6.1 xmc6.2 xmc7.0 xmc7.1
10 xmc7.2 xmc8.0 xmc8.1 xmc8.2 xmc9.0 xmc9.1 xmc9.2 xmc10.0
11 xmc10.1 xmc10.2 xmc11.0 xmc11.1 xmc11.2 xmc12.0 xmc12.1 xcm12.2
12 Nc1.0 Nc1.1 Nc1.2 Nc1.3 Nc1.4 Nc1.5 Nc1.6 bc1.0
13 bc1.1 Mc1.0 Mc1.1 xmaxc10 xmaxc11 xmaxc12 xmaxc13 xmaxc14
14 xmax15 xmc13.0 xmc13.1 xmc13.2 xmc14.0 xmc14.1 xmc14.2 xmc15.0
15 xmc15.1 xmc15.2 xmc16.0 xmc16.1 xmc16.2 xmc17.0 xmc17.1 xmc17.2
16 xmc18.0 xmc18.1 xmc18.2 xmc19.0 xmc19.1 xmc19.2 xmc20.0 xmc20.1
17 xmc20.2 xmc21.0 xmc21.1 xmc21.2 xmc22.0 xmc22.1 xmc22.2 xmc23.0
18 xmc23.1 xmc23.2 xmc24.0 xmc24.1 xmc24.2 xmc25.0 xmc25.1 xmc25.2
19 Nc2.0 Nc2.1 Nc2.2 Nc2.3 Nc2.4 Nc2.5 Nc2.6 bc2.0
20 bc2.1 Mc2.0 Mc2.1 xmaxc20 xmaxc21 xmaxc22 xmaxc23 xmaxc24
21 xmaxc25 xmc26.0 xmc26.1 xmc26.2 xmc27.0 xmc27.1 xmc27.2 xmc28.0
22 xmc28.1 xmc28.2 xmc29.0 xmc29.1 xmc29.2 xmc30.0 xmc30.1 xmc30.2
23 xmc31.0 xmc31.1 xmc31.2 xmc32.0 xmc32.1 xmc32.2 xmc33.0 xmc33.1
24 xmc33.2 xmc34.0 xmc34.1 xmc34.2 xmc35.0 xmc35.1 xmc35.2 xmc36.0
25 Xmc36.1 xmc36.2 xmc37.0 xmc37.1 xmc37.2 xmc38.0 xmc38.1 xmc38.2
26 Nc3.0 Nc3.1 Nc3.2 Nc3.3 Nc3.4 Nc3.5 Nc3.6 bc3.0
27 bc3.1 Mc3.0 Mc3.1 xmaxc30 xmaxc31 xmaxc32 xmaxc33 xmaxc34
28 xmaxc35 xmc39.0 xmc39.1 xmc39.2 xmc40.0 xmc40.1 xmc40.2 xmc41.0
29 xmc41.1 xmc41.2 xmc42.0 xmc42.1 xmc42.2 xmc43.0 xmc43.1 xmc43.2
30 xmc44.0 xmc44.1 xmc44.2 xmc45.0 xmc45.1 xmc45.2 xmc46.0 xmc46.1
31 xmc46.2 xmc47.0 xmc47.1 xmc47.2 xmc48.0 xmc48.1 xmc48.2 xmc49.0
32 xmc49.1 xmc49.2 xmc50.0 xmc50.1 xmc50.2 xmc51.0 xmc51.1 xmc51.2
Table 3: GSM payload format
Technical Specification TS 101 318.
4.5.12 GSM-EFR
GSM-EFR denotes GSM 06.60 enhanced full rate speech transcoding,
specified in ETS 300 969 which is available from ETSI at the address
given in Section 4.5.10. This codec has a frame length of 244 bits.
For transmission in RTP, each codec frame is packed into a 31 octet
(248 bit) buffer beginning with a 4-bit signature 0xC in a manner
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similar to that specified here for the original GSM 06.10 codec. The
packing is specified in ETSI Technical Specification TS 101 318.
4.5.13 L8
L8 denotes linear audio data samples, using 8-bits of precision with
an offset of 128, that is, the most negative signal is encoded as
zero.
4.5.14 L16
L16 denotes uncompressed audio data samples, using 16-bit signed
representation with 65535 equally divided steps between minimum and
maximum signal level, ranging from -32768 to 32767. The value is
represented in two's complement notation and transmitted in network
byte order (most significant byte first).
4.5.15 LPC
LPC designates an experimental linear predictive encoding contributed
by Ron Frederick, Xerox PARC, which is based on an implementation
written by Ron Zuckerman, Motorola, posted to the Usenet group
comp.dsp on June 26, 1992. The codec generates 14 octets for every
frame. The framesize is set to 20 ms, resulting in a bit rate of
5,600 b/s.
4.5.16 MPA
MPA denotes MPEG-1 or MPEG-2 audio encapsulated as elementary
streams. The encoding is defined in ISO standards ISO/IEC 11172-3
and 13818-3. The encapsulation is specified in RFC 2250 [16].
Sampling rate and channel count are contained in the payload. MPEG-1
audio supports sampling rates of 32, 44.1, and 48 kHz (ISO/IEC
11172-3, section 1.1; "Scope"). MPEG-2 additionally supports sampling
rates of 16, 22.05 and 24 kHz.
4.5.17 PCMA and PCMU
PCMA and PCMU are specified in ITU-T Recommendation G.711. Audio data
is encoded as eight bits per sample, after logarithmic scaling. PCMU
denotes mu-law scaling, PCMA A-law scaling. A detailed description is
given by Jayant and Noll [17]. Each G.711 octet SHALL be octet-
aligned in an RTP packet. The sign bit of each G.711 octet SHALL
correspond to the most significant bit of the octet in the RTP packet
(i.e., assuming the G.711 samples are handled as octets on the host
machine, the sign bit SHALL be the most signficant bit of the octet
as defined by the host machine format). The 56 kb/s and 48 kb/s modes
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of G.711 are not applicable to RTP, since PCMA and PCMU SHALL always
be transmitted as 8-bit samples.
4.5.18 QCELP
The Electronic Industries Association (EIA) & Telecommunications
Industry Association (TIA) standard IS-733, "TR45: High Rate Speech
Service Option for Wideband Spread Spectrum Communications Systems,"
defines the QCELP audio compression algorithm for use in wireless
CDMA applications. The QCELP CODEC compresses each 20 milliseconds of
8000 Hz, 16- bit sampled input speech into one of four different size
output frames: Rate 1 (266 bits), Rate 1/2 (124 bits), Rate 1/4 (54
bits) or Rate 1/8 (20 bits). For typical speech patterns, this
results in an average output of 6.8 k bits/sec for normal mode and
4.7 k bits/sec for reduced rate mode. The packetization of the QCELP
audio codec is described in [18].
4.5.19 RED
The redundant audio payload format "RED" is specified by RFC 2198
[19]. It defines a means by which multiple redundant copies of an
audio packet may be transmitted in a single RTP stream. Each packet
in such a stream contains, in addition to the audio data for that
packetization interval, a (more heavily compressed) copy of the data
from a previous packetization interval. This allows an approximation
of the data from lost packets to be recovered upon decoding of a
subsequent packet, giving much improved sound quality when compared
with silence substitution for lost packets.
4.5.20 SX*
The SX7300P, SX8300P and SX9600P codecs are part of the same
compatible family and distinguished by the first octet in each frame,
where "x" can be any value:
0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|0 0 x | SX7300P bitstream (14 byte frame)
|0 1 0 | SX8300P bitstream (16 byte frame)
|1 0 x | VAD bistream ( 2 byte frame)
|1 1 x | SX9600P bitstream (18 byte frame)
+-+-+-+-+-+-+-+-+
4.5.20.1 SX7300P
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The SX7300P is a low-complexity CELP-based audio codec operating at a
sampling rate of 8000 Hz. It encodes blocks of 120 audio samples (15
ms) into an encoded frame of 14 octets, yielding an encoded bit rate
of approximately 7467 b/s.
4.5.20.2 SX8300P
The SX8300P is a low-complexity CELP-based audio codec operating at a
sampling rate of 8000 Hz. It encodes blocks of 120 audio samples (15
ms) into an encoded frame of 16 octets, yielding an encoded bit rate
of approximately 8533 b/s.
4.5.20.3 SX9600P
The SX9600P is a low-complexity, toll-quality CELP-based audio codec
operating at a sampling rate of 8000 Hz. It encodes blocks of 120
audio samples (15 ms) into an encoded frame of 18 octets, yielding an
encoded bit rate of 9600 b/s.
4.5.21 VDVI
VDVI is a variable-rate version of DVI4, yielding speech bit rates of
between 10 and 25 kb/s. It is specified for single-channel operation
only. Samples are packed into octets starting at the most-
significant bit.
It uses the following encoding:
DVI4 codeword VDVI bit pattern
_________________________________
0 00
1 010
2 1100
3 11100
4 111100
5 1111100
6 11111100
7 11111110
8 10
9 011
10 1101
11 11101
12 111101
13 1111101
14 11111101
15 11111111
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5 Video
The following sections describe the video encodings that are defined
in this memo and give their abbreviated names used for
identification. These video encodings and their payload types are
listed in Table 5.
All of these video encodings use an RTP timestamp frequency of 90,000
Hz, the same as the MPEG presentation time stamp frequency. This
frequency yields exact integer timestamp increments for the typical
24 (HDTV), 25 (PAL), and 29.97 (NTSC) and 30 Hz (HDTV) frame rates
and 50, 59.94 and 60 Hz field rates. While 90 kHz is the RECOMMENDED
rate for future video encodings used within this profile, other rates
MAY be used. However, it is not sufficient to use the video frame
rate (typically between 15 and 30 Hz) because that does not provide
adequate resolution for typical synchronization requirements when
calculating the RTP timestamp corresponding to the NTP timestamp in
an RTCP SR packet. The timestamp resolution MUST also be sufficient
for the jitter estimate contained in the receiver reports.
For most of these video encodings, the RTP timestamp encodes the
sampling instant of the video image contained in the RTP data packet.
If a video image occupies more than one packet, the timestamp is the
same on all of those packets. Packets from different video images are
distinguished by their different timestamps.
Most of these video encodings also specify that the marker bit of the
RTP header SHOULD be set to one in the last packet of a video frame
and otherwise set to zero. Thus, it is not necessary to wait for a
following packet with a different timestamp to detect that a new
frame should be displayed.
5.1 BT656
The encoding is specified in ITU-R Recommendation BT.656-3,
"Interfaces for Digital Component Video Signals in 525-Line and 625-
Line Television Systems operating at the 4:2:2 Level of
Recommendation ITU-R BT.601 (Part A)". The packetization and RTP-
specific properties are described in RFC 2431 [20].
5.2 CelB
The CELL-B encoding is a proprietary encoding proposed by Sun
Microsystems. The byte stream format is described in RFC 2029 [21].
5.3 JPEG
The encoding is specified in ISO Standards 10918-1 and 10918-2. The
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RTP payload format is as specified in RFC 2435 [22].
5.4 H261
The encoding is specified in ITU-T Recommendation H.261, "Video codec
for audiovisual services at p x 64 kbit/s". The packetization and
RTP-specific properties are described in RFC 2032 [23].
5.5 H263
The encoding is specified in the 1996 version of ITU-T Recommendation
H.263, "Video coding for low bit rate communication". The
packetization and RTP-specific properties are described in RFC 2190
[24].
5.6 H263-1998
The encoding is specified in the 1998 version of ITU-T Recommendation
H.263, "Video coding for low bit rate communication". The
packetization and RTP-specific properties are described in RFC 2429
[25]. Because the 1998 version of H.263 is a superset of the 1996
syntax, this payload format can also be used with the 1996 version of
H.263, and is recommended for this use by new implementations. This
payload format does not replace RFC 2190, which continues to be used
by existing implementations, and may be required for backward
compatibility in new implementations. Implementations using the new
features of the 1998 version of H.263 MUST use the payload format
described in RFC 2429.
5.7 MPV
MPV designates the use of MPEG-1 and MPEG-2 video encoding elementary
streams as specified in ISO Standards ISO/IEC 11172 and 13818-2,
respectively. The RTP payload format is as specified in RFC 2250
[16], Section 3.
5.8 MP2T
MP2T designates the use of MPEG-2 transport streams, for either audio
or video. The RTP payoad format is described in RFC 2250 [16],
Section 2.
5.9 MP1S
MP1S designates an MPEG-1 systems stream, encapsulated according to
RFC 2250 [16].
5.10 MP2P
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MP2P designates an MPEG-2 program stream, encapsulated according to
RFC 2250 [16].
5.11 BMPEG
BMPEG designates an experimental payload format for MPEG-1 and MPEG-2
which specifies bundled (multiplexed) transport of audio and video
elementary streams in one RTP stream as an alternative to the MP1S
and MP2P formats. The packetization is described in RFC 2343 [26].
5.12 nv
The encoding is implemented in the program `nv', version 4, developed
at Xerox PARC by Ron Frederick. Further information is available from
the author:
Ron Frederick
Xerox Palo Alto Research Center
3333 Coyote Hill Road
Palo Alto, CA 94304
United States
electronic mail: frederic@parc.xerox.com
6 Payload Type Definitions
Tables 4 and 5 define this profile's static payload type values for
the PT field of the RTP data header. In addition, payload type
values in the range 96-127 MAY be defined dynamically through a
conference control protocol, which is beyond the scope of this
document. For example, a session directory could specify that for a
given session, payload type 96 indicates PCMU encoding, 8,000 Hz
sampling rate, 2 channels. Entries in Tables 4 and 5 with payload
type "dyn" have no static payload type assigned and are only used
with a dynamic payload type. The payload type range marked `reserved'
has been set aside so that RTCP and RTP packets can be reliably
distinguished (see Section "Summary of Protocol Constants" of the RTP
protocol specification).
The payload types currently defined in this profile are assigned to
exactly one of three categories or media types : audio only, video
only and those combining audio and video. The media types are marked
in Tables 4 and 5 as "A", "V" and "AV", respectively. Payload types
of different media types SHALL NOT be interleaved or multiplexed
within a single RTP session, but multiple RTP sessions MAY be used in
parallel to send multiple media types. An RTP source MAY change
payload types within the same media type during a session.
Session participants agree through mechanisms beyond the scope of
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this specification on the set of payload types allowed in a given
session. This set MAY, for example, be defined by the capabilities
of the applications used, negotiated by a conference control protocol
or established by agreement between the human participants.
Audio applications operating under this profile SHOULD, at a minimum,
be able to send and/or receive payload types 0 (PCMU) and 5 (DVI4).
This allows interoperability without format negotiation and ensures
successful negotation with a conference control protocol.
PT encoding media type clock rate channels
name (Hz)
___________________________________________________________
0 PCMU A 8000 1
1 1016 A 8000 1
2 G726-32 A 8000 1
3 GSM A 8000 1
4 G723 A 8000 1
5 DVI4 A 8000 1
6 DVI4 A 16000 1
7 LPC A 8000 1
8 PCMA A 8000 1
9 G722 A 16000 1
10 L16 A 44100 2
11 L16 A 44100 1
12 QCELP A 8000 1
13 unassigned A
14 MPA A 90000 (see text)
15 G728 A 8000 1
16 DVI4 A 11025 1
17 DVI4 A 22050 1
18 G729 A 8000 1
19 CN A 8000 1
20 unassigned A
21 unassigned A
22 unassigned A
23 unassigned A
dyn GSM-HR A 8000 1
dyn GSM-EFR A 8000 1
dyn RED A
Table 4: Payload types (PT) for audio encodings
7 RTP over TCP and Similar Byte Stream Protocols
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PT encoding media type clock rate
name (Hz)
____________________________________________________
24 unassigned V
25 CelB V 90000
26 JPEG V 90000
27 unassigned V
28 nv V 90000
29 unassigned V
30 unassigned V
31 H261 V 90000
32 MPV V 90000
33 MP2T AV 90000
34 H263 V 90000
35-71 unassigned ?
72-76 reserved N/A N/A
77-95 unassigned ?
96-127 dynamic ?
dyn BT656 V 90000
dyn H263-1998 V 90000
dyn MP1S V 90000
dyn MP2P V 90000
dyn BMPEG V 90000
Table 5: Payload types (PT) for video and combined encodings
Under special circumstances, it may be necessary to carry RTP in
protocols offering a byte stream abstraction, such as TCP, possibly
multiplexed with other data. If the application does not define its
own method of delineating RTP and RTCP packets, it SHOULD prefix each
packet with a two-octet length field.
(Note: RTSP [27] provides its own encapsulation and does not need an
extra length indication.)
8 Port Assignment
As specified in the RTP protocol definition, RTP data SHOULD be
carried on an even UDP or TCP port number and the corresponding RTCP
packets SHOULD be carried on the next higher (odd) port number.
Applications operating under this profile MAY use any such UDP or TCP
port pair. For example, the port pair MAY be allocated randomly by a
session management program. A single fixed port number pair cannot be
required because multiple applications using this profile are likely
to run on the same host, and there are some operating systems that do
not allow multiple processes to use the same UDP port with different
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multicast addresses.
However, port numbers 5004 and 5005 have been registered for use with
this profile for those applications that choose to use them as the
default pair. Applications that operate under multiple profiles MAY
use this port pair as an indication to select this profile if they
are not subject to the constraint of the previous paragraph.
Applications need not have a default and MAY require that the port
pair be explicitly specified. The particular port numbers were chosen
to lie in the range above 5000 to accommodate port number allocation
practice within some versions of the Unix operating system, where
port numbers below 1024 can only be used by privileged processes and
port numbers between 1024 and 5000 are automatically assigned by the
operating system.
9 Changes from RFC 1890
This RFC revises RFC 1890. It is fully backwards-compatible with RFC
1890 and codifies existing practice. The changes are listed below.
o Additional payload formats and/or expanded descriptions were
included for CN, G723, G726, G727, G728, G729, GSM, GSM-HR,
GSM-EFR, QCELP, RED, BT656, H263-1998, MP1S, MP2P and BMPEG.
o Static payload types 4, 12, 16, 17, 18, 19 and 34 were added.
o The policy is established that no additional registration of
static payload types for this Profile will be made beyond those
included in Tables 4 and 5, but additional encoding names may
be registered as MIME subtypes.
o In Section 4.1, the requirement level for setting of the
marker bit on the first packet after silence for audio was
changed from "is" to "SHOULD be".
o Similarly, text was added to specify that the marker bit
SHOULD be set to one on the last packet of a video frame, and
that video frames are distinguished by their timestamps.
o This profile follows the suggestion in the RTP spec that RTCP
bandwidth may be specified separately from the session
bandwidth and separately for active senders and passive
receivers.
o RFC references are added for payload formats published after
RFC 1890.
o A minimal TCP encapsulation is defined.
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o The security considerations and full copyright sections were
added.
o According to Peter Hoddie of Apple, only pre-1994 Macintosh
used the 22254.54 rate and none the 11127.27 rate, so the
latter was dropped from the discussion of suggested sampling
frequencies.
o Small clarifications of the text have been made in several
places, some in response to questions from readers. In
particular:
-A definition for "media type" is given in Section 1.1 to allow
the explanation of multiplexing RTP sessions in Section 6 to
be more clear regarding the multiplexing of multiple media.
-The explanation of how to determine the number of audio frames
in a packet from the length was expanded.
-More description of the allocation of bandwidth to SDES items
is given.
-The terms MUST, SHOULD, MAY, etc. are used as defined in RFC
2119.
10 Security Considerations
Implementations using the profile defined in this specification are
subject to the security considerations discussed in the RTP
specification [1]. This profile does not specify any different
security services other than giving rules for mapping characters in a
user-provided pass phrase to canonical form. The primary function of
this profile is to list a set of data compression encodings for audio
and video media.
Confidentiality of the media streams is achieved by encryption.
Because the data compression used with the payload formats described
in this profile is applied end-to-end, encryption may be performed
after compression so there is no conflict between the two operations.
A potential denial-of-service threat exists for data encodings using
compression techniques that have non-uniform receiver-end
computational load. The attacker can inject pathological datagrams
into the stream which are complex to decode and cause the receiver to
be overloaded. However, the encodings described in this profile do
not exhibit any significant non-uniformity.
As with any IP-based protocol, in some circumstances a receiver may
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be overloaded simply by the receipt of too many packets, either
desired or undesired. Network-layer authentication may be used to
discard packets from undesired sources, but the processing cost of
the authentication itself may be too high. In a multicast
environment, pruning of specific sources may be implemented in future
versions of IGMP [28] and in multicast routing protocols to allow a
receiver to select which sources are allowed to reach it.
11 Full Copyright Statement
Copyright (C) The Internet Society (1999). All Rights Reserved.
This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it
or assist in its implmentation may be prepared, copied, published and
distributed, in whole or in part, without restriction of any kind,
provided that the above copyright notice and this paragraph are
included on all such copies and derivative works. However, this
document itself may not be modified in any way, such as by removing
the copyright notice or references to the Internet Society or other
Internet organizations, except as needed for the purpose of
developing Internet standards in which case the procedures for
copyrights defined in the Internet Standards process must be
followed, or as required to translate it into languages other than
English.
The limited permissions granted above are perpetual and will not be
revoked by the Internet Society or its successors or assigns.
This document and the information contained herein is provided on an
"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
12 Acknowledgements
The comments and careful review of Steve Casner, Simao Campos and
Richard Cox are gratefully acknowledged. The GSM description was
adopted from the IMTC Voice over IP Forum Service Interoperability
Implementation Agreement (January 1997). Fred Burg and Terry Lyons
helped with the G.729 description.
13 Address of Author
Henning Schulzrinne
Dept. of Computer Science
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Columbia University
1214 Amsterdam Avenue
New York, NY 10027
USA
electronic mail: schulzrinne@cs.columbia.edu
A Bibliography
[1] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: A
transport protocol for real-time applications," Internet Draft,
Internet Engineering Task Force, Feb. 1999 Work in progress, revision
to RFC 1889.
[2] S. Bradner, "Key words for use in RFCs to Indicate Requirement
Levels," RFC 2119, Internet Engineering Task Force, Mar. 1997.
[3] N. Freed, J. Klensin, and J. Postel, "Multipurpose Internet Mail
Extensions (MIME) Part Four: Registration Procedures," RFC 2048,
Internet Engineering Task Force, Nov. 1996.
[4] P. Hoschka, "MIME Type Registration of RTP Payload Types,"
Internet Draft, Internet Engineering Task Force, Feb. 1999 Work in
progress.
[5] M. Handley and V. Jacobson, "SDP: Session Description Protocol,"
Request for Comments (Proposed Standard) RFC 2327, Internet
Engineering Task Force, Apr. 1998.
[6] Apple Computer, "Audio interchange file format AIFF-C," Aug.
1991. (also ftp://ftp.sgi.com/sgi/aiff-c.9.26.91.ps.Z).
[7] Office of Technology and Standards, "Telecommunications: Analog
to digital conversion of radio voice by 4,800 bit/second code excited
linear prediction (celp)," Federal Standard FS-1016, GSA, Room 6654;
7th & D Street SW; Washington, DC 20407 (+1-202-708-9205), 1990.
[8] J. P. Campbell, Jr., T. E. Tremain, and V. C. Welch, "The
proposed Federal Standard 1016 4800 bps voice coder: CELP," Speech
Technology , vol. 5, pp. 58--64, April/May 1990.
[9] J. P. Campbell, Jr., T. E. Tremain, and V. C. Welch, "The federal
standard 1016 4800 bps CELP voice coder," Digital Signal Processing ,
vol. 1, no. 3, pp. 145--155, 1991.
[10] J. P. Campbell, Jr., T. E. Tremain, and V. C. Welch, "The DoD
4.8 kbps standard (proposed federal standard 1016)," in Advances in
Speech Coding (B. Atal, V. Cuperman, and A. Gersho, eds.), ch. 12,
pp. 121--133, Kluwer Academic Publishers, 1991.
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[11] IMA Digital Audio Focus and Technical Working Groups,
"Recommended practices for enhancing digital audio compatibility in
multimedia systems (version 3.00)," tech. rep., Interactive
Multimedia Association, Annapolis, Maryland, Oct. 1992.
[12] D. Deleam and J.-P. Petit, "Real-time implementations of the
recent ITU-T low bit rate speech coders on the TI TMS320C54X DSP:
results, methodology, and applications," in Proc. of International
Conference on Signal Processing, Technology, and Applications
(ICSPAT) , (Boston, Massachusetts), pp. 1656--1660, Oct. 1996.
[13] M. Mouly and M.-B. Pautet, The GSM system for mobile
communications Lassay-les-Chateaux, France: Europe Media Duplication,
1993.
[14] J. Degener, "Digital speech compression," Dr. Dobb's Journal ,
Dec. 1994.
[15] S. M. Redl, M. K. Weber, and M. W. Oliphant, An Introduction to
GSM Boston: Artech House, 1995.
[16] D. Hoffman, G. Fernando, V. Goyal, and M. Civanlar, "RTP payload
format for MPEG1/MPEG2 video," Request for Comments (Proposed
Standard) RFC 2250, Internet Engineering Task Force, Jan. 1998.
[17] N. S. Jayant and P. Noll, Digital Coding of Waveforms--
Principles and Applications to Speech and Video Englewood Cliffs, New
Jersey: Prentice-Hall, 1984.
[18] K. McKay, "RTP Payload Format for PureVoice(tm) Audio", Internet
Draft, Internet Engineering Task Force, Oct. 1998. Work in progress.
[19] C. Perkins, I. Kouvelas, O. Hodson, V. Hardman, M. Handley, J.C.
Bolot, A. Vega-Garcia, and S. Fosse-Parisis, "RTP Payload for
Redundant Audio Data," Request for Comments (Proposed Standard) RFC
2198, Internet Engineering Task Force, Sep. 1997.
[20] D. Tynan, "RTP payload format for BT.656 Video Encoding,"
Request for Comments (Proposed Standard) RFC 2431, Internet
Engineering Task Force, Oct. 1998.
[21] M. Speer and D. Hoffman, "RTP payload format of sun's CellB
video encoding," Request for Comments (Proposed Standard) RFC 2029,
Internet Engineering Task Force, Oct. 1996.
[22] L. Berc, W. Fenner, R. Frederick, and S. McCanne, "RTP payload
format for JPEG-compressed video," Request for Comments (Proposed
Standard) RFC 2435, Internet Engineering Task Force, Oct. 1996.
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[23] T. Turletti and C. Huitema, "RTP payload format for H.261 video
streams," Request for Comments (Proposed Standard) RFC 2032, Internet
Engineering Task Force, Oct. 1996.
[24] C. Zhu, "RTP payload format for H.263 video streams," Request
for Comments (Proposed Standard) RFC 2190, Internet Engineering Task
Force, Sep. 1997.
[25] C. Bormann, L. Cline, G. Deisher, T. Gardos, C. Maciocco, D.
Newell, J. Ott, G. Sullivan, S. Wenger, C. Zhu, "RTP Payload Format
for the 1998 Version of ITU-T Rec. H.263 Video (H.263+)," Request for
Comments (Proposed Standard) RFC 2429, Internet Engineering Task
Force, Oct. 1998.
[26] M. Civanlar, G. Cash, B. Haskell, "RTP Payload Format for
Bundled MPEG," Request for Comments (Experimental) RFC 2343, Internet
Engineering Task Force, May 1998.
[27] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming
protocol (RTSP)," Request for Comments (Proposed Standard) RFC 2326,
Internet Engineering Task Force, Apr. 1998.
[28] S. Deering, "Host Extensions for IP Multicasting," Request for
Comments RFC 1112, STD 5, Internet Engineering Task Force, Aug. 1989.
Current Locations of Related Resources
Note: Several sections below refer to the ITU-T Software Tool Library
(STL). It is available from the ITU Sales Service, Place des Nations,
CH-1211 Geneve 20, Switzerland (also check http://www.itu.int. The
ITU-T STL is covered by a license defined in ITU-T Recommendation
G.191, "Software tools for speech and audio coding standardization".
UTF-8
Information on the UCS Transformation Format 8 (UTF-8) is available
at
http://www.stonehand.com/unicode/standard/utf8.html
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
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no charge (on DOS diskettes, but configured to compile on Sun SPARC
stations) from: Bob Fenichel, National Communications System,
Washington, D.C. 20305, phone +1-703-692-2124, fax +1-703-746-4960.
An implementation is also available at
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
G722
An implementation of the G.722 algorithm is available as part of the
ITU-T STL, described above.
G723
The reference C code implementation defining the G.723.1 algorithm
and its Annexes A, B, and C are available as an integral part of
Recommendation G.723.1 from the ITU Sales Service, address listed
above. Both the algorithm and C code are covered by a specific
license. The ITU-T Secretariat should be contacted to obtain such
licensing information.
G726-16 through G726-40
G726-16 through G726-40 are specified in the ITU-T Recommendation
G.726, "40, 32, 24, and 16 kb/s Adaptive Differential Pulse Code
Modulation (ADPCM)". An implementation of the G.726 algorithm is
available as part of the ITU-T STL, described above.
G727-16 through G727-40
G727-16 through G727-40 are specified in the ITU-T Recommendation
G.727, "5-, 4-, 3-, and 2-bit/sample embedded adaptive differential
pulse code modulation". An implementation of the G.727 algorithm will
be available in a future release of the ITU-T STL, described above.
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G729
The reference C code implementation defining the G.729 algorithm and
its Annexes A and B are available as an integral part of
Recommendation G.729 from the ITU Sales Service, listed above. Both
the algorithm and the C code are covered by a specific license. The
contact information for obtaining the license is listed in the C
code.
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/
Although the RPE-LTP algorithm is not an ITU-T standard, there is a C
code implementation of the RPE-LTP algorithm available as part of the
ITU-T STL. The STL implementation is an adaptation of the TU Berlin
version.
LPC
An implementation is available at
ftp://parcftp.xerox.com/pub/net-research/lpc.tar.Z
PCMU, PCMA
An implementation of these algorithm is available as part of the
ITU-T STL, described above. Code to convert between linear and mu-law
companded data is also available in [11].
Table of Contents
1 Introduction ........................................ 2
1.1 Terminology ......................................... 3
2 RTP and RTCP Packet Forms and Protocol Behavior ..... 3
3 Registering Additional Encodings with IANA .......... 6
4 Audio ............................................... 7
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4.1 Encoding-Independent Rules .......................... 7
4.2 Operating Recommendations ........................... 9
4.3 Guidelines for Sample-Based Audio Encodings ......... 9
4.4 Guidelines for Frame-Based Audio Encodings .......... 10
4.5 Audio Encodings ..................................... 10
4.5.1 1016 ................................................ 12
4.5.2 CN .................................................. 12
4.5.3 DVI4 ................................................ 13
4.5.4 G722 ................................................ 14
4.5.5 G723 ................................................ 14
4.5.6 G726-16, G726-24, G726-32, G726-40 .................. 15
4.5.7 G727-16, G727-24, G727-32, G727-40 .................. 15
4.5.8 G728 ................................................ 15
4.5.9 G729 ................................................ 16
4.5.10 GSM ................................................. 18
4.5.10.1 General Packaging Issues ............................ 18
4.5.10.2 GSM variable names and numbers ...................... 18
4.5.11 GSM-HR .............................................. 18
4.5.12 GSM-EFR ............................................. 20
4.5.13 L8 .................................................. 21
4.5.14 L16 ................................................. 21
4.5.15 LPC ................................................. 21
4.5.16 MPA ................................................. 21
4.5.17 PCMA and PCMU ....................................... 21
4.5.18 QCELP ............................................... 22
4.5.19 RED ................................................. 22
4.5.20 SX* ................................................. 22
4.5.20.1 SX7300P ............................................. 22
4.5.20.2 SX8300P ............................................. 23
4.5.20.3 SX9600P ............................................. 23
4.5.21 VDVI ................................................ 23
5 Video ............................................... 24
5.1 BT656 ............................................... 24
5.2 CelB ................................................ 24
5.3 JPEG ................................................ 24
5.4 H261 ................................................ 25
5.5 H263 ................................................ 25
5.6 H263-1998 ........................................... 25
5.7 MPV ................................................. 25
5.8 MP2T ................................................ 25
5.9 MP1S ................................................ 25
5.10 MP2P ................................................ 25
5.11 BMPEG ............................................... 26
5.12 nv .................................................. 26
6 Payload Type Definitions ............................ 26
7 RTP over TCP and Similar Byte Stream Protocols ...... 27
8 Port Assignment ..................................... 28
9 Changes from RFC 1890 ............................... 29
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10 Security Considerations ............................. 30
11 Full Copyright Statement ............................ 31
12 Acknowledgements .................................... 31
13 Address of Author ................................... 31
A Bibliography ........................................ 32
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