Network Working Group H. Schulzrinne
Request for Comments: 3551 Columbia University
Obsoletes: 1890 S. Casner
Category: Standards Track Packet Design
July 2003 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.
Copyright (C) The Internet Society (2003). All Rights Reserved.
This document describes a profile called "RTP/AVP" for the use of the
real-time transport protocol (RTP), version 2, and the associated
control protocol, RTCP, within audio and video multiparticipant
conferences with minimal control. It provides interpretations of
generic fields within the RTP specification suitable for audio and
video conferences. In particular, this document defines a set of
default mappings from payload type numbers to encodings.
This document also describes how audio and video data may be carried
within RTP. It defines a set of standard encodings and their names
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
This memorandum obsoletes RFC 1890. It is mostly backwards-
compatible except for functions removed because two interoperable
implementations were not found. The additions to RFC 1890 codify
existing practice in the use of payload formats under this profile
and include new payload formats defined since RFC 1890 was published.
14. Acknowledgments .............................................. 4215. Intellectual Property Rights Statement ....................... 4316. Authors' Addresses ........................................... 4317. Full Copyright Statement ..................................... 441. Introduction
This profile defines aspects of RTP left unspecified in the RTP
Version 2 protocol definition (RFC 3550) . 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 may use the name for this profile specified in Section
Other profiles may make different choices for the items specified
This document also defines a set of encodings and payload formats for
audio and video. These payload format descriptions are included here
only as a matter of convenience since they are too small to warrant
separate documents. Use of these payload formats is NOT REQUIRED to
use this profile. Only the binding of some of the payload formats to
static payload type numbers in Tables 4 and 5 is normative.
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  and
indicate requirement levels for implementations compliant with this
This document defines the term media type as dividing encodings of
audio and video content into three classes: audio, video and
2. RTP and RTCP Packet Forms and Protocol Behavior
The section "RTP Profiles and Payload Format Specifications" of RFC
3550 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. 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  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: No mapping is specified by this profile.
Congestion: RTP and this profile may be used in the context of
enhanced network service, for example, through Integrated Services
(RFC 1633)  or Differentiated Services (RFC 2475) , or they
may be used with best effort service.
If enhanced service is being used, RTP receivers SHOULD monitor
packet loss to ensure that the service that was requested is
actually being delivered. If it is not, then they SHOULD assume
that they are receiving best-effort service and behave
If best-effort service is being used, RTP receivers SHOULD monitor
packet loss to ensure that the packet loss rate is within
acceptable parameters. Packet loss is considered acceptable if a
TCP flow across the same network path and experiencing the same
network conditions would achieve an average throughput, measured
on a reasonable timescale, that is not less than the RTP flow is
achieving. This condition can be satisfied by implementing
congestion control mechanisms to adapt the transmission rate (or
the number of layers subscribed for a layered multicast session),
or by arranging for a receiver to leave the session if the loss
rate is unacceptably high.
The comparison to TCP cannot be specified exactly, but is intended
as an "order-of-magnitude" comparison in timescale and throughput.
The timescale on which TCP throughput is measured is the round-
trip time of the connection. In essence, this requirement states
that it is not acceptable to deploy an application (using RTP or
any other transport protocol) on the best-effort Internet which
consumes bandwidth arbitrarily and does not compete fairly with
TCP within an order of magnitude.
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-
Transport mapping: The standard mapping of RTP and RTCP to
transport-level addresses is used.
Encapsulation: This profile leaves to applications the
specification of RTP encapsulation in protocols other than UDP.
3. Registering Additional Encodings
This profile lists a set of encodings, each of which is comprised of
a particular media data compression or representation plus a payload
format for encapsulation within RTP. Some of those payload formats
are specified here, while others are specified in separate RFCs. It
is expected that additional encodings beyond the set listed here will
be created in the future and specified in additional payload format
This profile also assigns to each encoding a short name which MAY be
used by higher-level control protocols, such as the Session
Description Protocol (SDP), RFC 2327 , to identify encodings
selected for a particular RTP session.
In some contexts it may be useful to refer to these encodings in the
form of a MIME content-type. To facilitate this, RFC 3555 
provides registrations for all of the encodings names listed here as
MIME subtype names under the "audio" and "video" MIME types through
the MIME registration procedure as specified in RFC 2048 .
Any additional encodings specified for use under this profile (or
others) may also be assigned names registered as MIME subtypes with
the Internet Assigned Numbers Authority (IANA). This registry
provides a means to insure that the names assigned to the additional
encodings are kept unique. RFC 3555 specifies the information that
is required for the registration of RTP encodings.
In addition to assigning names to encodings, this profile also
assigns static RTP payload type numbers to some of them. However,
the payload type number space is relatively small and cannot
accommodate assignments for all existing and future encodings.
During the early stages of RTP development, it was necessary to use
statically assigned payload types because no other mechanism had been
specified to bind encodings to payload types. It was anticipated
that non-RTP means beyond the scope of this memo (such as directory
services or invitation protocols) would be specified to establish a
dynamic mapping between a payload type and an encoding. Now,
mechanisms for defining dynamic payload type bindings have been
specified in the Session Description Protocol (SDP) and in other
protocols such as 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, with 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. Those 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
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
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
The final set of static payload type assignments is provided in
Tables 4 and 5.
4.1 Encoding-Independent Rules
Since the ability to suppress silence is one of the primary
motivations for using packets to transmit voice, the RTP header
carries both a sequence number and a timestamp to allow a receiver to
distinguish between lost packets and periods of time when no data was
transmitted. Discontiguous transmission (silence suppression) MAY be
used with any audio payload format. Receivers MUST assume that
senders may suppress silence unless this is restricted by signaling
specified elsewhere. (Even if the transmitter does not suppress
silence, the receiver should be prepared to handle periods when no
data is present since packets may be lost.)
Some payload formats (see Sections 4.5.3 and 4.5.6) define a "silence
insertion descriptor" or "comfort noise" frame to specify parameters
for artificial noise that may be generated during a period of silence
to approximate the background noise at the source. For other payload
formats, a generic Comfort Noise (CN) payload format is specified in
RFC 3389 . When the CN payload format is used with another
payload format, different values in the RTP payload type field
distinguish comfort-noise packets from those of the selected payload
For applications which send either no packets or occasional comfort-
noise packets during silence, the first packet of a talkspurt, that
is, the first packet after a silence period during which packets have
not been transmitted contiguously, SHOULD be distinguished by setting
the marker bit in the RTP data header to one. The marker bit 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
The RTP clock rate used for generating the RTP timestamp is
independent of the number of channels and the encoding; it usually
equals the number of sampling periods per second. For N-channel
encodings, each sampling period (say, 1/8,000 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 , using the following
notation, unless some other convention is specified for a particular
encoding or payload format:
channels description channel
1 2 3 4 5 6
2 stereo l r
3 l r c
4 l c r S
5 Fl Fr Fc Sl Sr
6 l lc c r rc S
Note: RFC 1890 defined two conventions for the ordering of four
audio channels. Since the ordering is indicated implicitly by
the number of channels, this was ambiguous. In this revision,
the order described as "quadrophonic" has been eliminated to
remove the ambiguity. This choice was based on the observation
that quadrophonic consumer audio format did not become popular
whereas surround-sound subsequently has.
Samples for all channels belonging to a single sampling instant MUST
be within the same packet. The interleaving of samples from
different channels depends on the encoding. General guidelines are
given in Section 4.3 and 4.4.
The sampling frequency SHOULD be drawn from the set: 8,000, 11,025,
16,000, 22,050, 24,000, 32,000, 44,100 and 48,000 Hz. (Older Apple
Macintosh computers had a native sample rate of 22,254.54 Hz, which
can be converted to 22,050 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
For packetized audio, the default packetization interval SHOULD have
a duration of 20 ms or one frame, whichever is longer, unless
otherwise noted in Table 1 (column "ms/packet"). The packetization
interval determines the minimum end-to-end delay; longer packets
introduce less header overhead but higher delay and make packet loss
more noticeable. For non-interactive applications such as lectures
or for links with severe bandwidth constraints, a higher
packetization delay MAY be used. A receiver SHOULD accept packets
representing between 0 and 200 ms of audio data. (For framed audio
encodings, a receiver SHOULD accept packets with a number of frames
equal to 200 ms divided by the frame duration, rounded up.) This
restriction allows reasonable buffer sizing for the receiver.
4.3 Guidelines for Sample-Based Audio Encodings
In sample-based encodings, each audio sample is represented by a
fixed number of bits. Within the compressed audio data, codes for
individual samples may span octet boundaries. An RTP audio packet
may contain any number of audio samples, subject to the constraint
that the number of bits per sample times the number of samples per
packet yields an integral octet count. Fractional encodings produce
less than one octet per sample.
The duration of an audio packet is determined by the number of
samples in the packet.
For sample-based encodings producing one or more octets per sample,
samples from different channels sampled at the same sampling instant
SHOULD be packed in consecutive octets. For example, for a two-
channel encoding, the octet sequence is (left channel, first sample),
(right channel, first sample), (left channel, second sample), (right
channel, second sample), .... For multi-octet encodings, octets
SHOULD be transmitted in network byte order (i.e., most significant
The packing of sample-based encodings producing less than one octet
per sample is encoding-specific.
The RTP timestamp reflects the instant at which the first sample in
the packet was sampled, that is, the oldest information in the
4.4 Guidelines for Frame-Based Audio Encodings
Frame-based encodings encode a fixed-length block of audio into
another block of compressed data, typically also of fixed length.
For frame-based encodings, the sender MAY choose to combine several
such frames into a single RTP packet. The receiver can tell the
number of frames contained in an RTP packet, if all the frames have
the same length, by dividing the RTP payload length by the audio
frame size which is defined as part of the encoding. This does not
work when carrying frames of different sizes unless the frame sizes
are relatively prime. If not, the frames MUST indicate their size.
For frame-based codecs, the channel order is defined for the whole
block. That is, for two-channel audio, right and left samples SHOULD
be coded independently, with the encoded frame for the left channel
preceding that for the right channel.
All frame-oriented audio codecs SHOULD be able to encode and decode
several consecutive frames within a single packet. Since the frame
size for the frame-oriented codecs is given, there is no need to use
a separate designation for the same encoding, but with different
number of frames per packet.
RTP packets SHALL contain a whole number of frames, with frames
inserted according to age within a packet, so that the oldest frame
(to be played first) occurs immediately after the RTP packet header.
The RTP timestamp reflects the instant at which the first sample in
the first frame was sampled, that is, the oldest information in the
4.5 Audio Encodings
name of sampling default
encoding sample/frame bits/sample rate ms/frame ms/packet
DVI4 sample 4 var. 20
G722 sample 8 16,000 20
G723 frame N/A 8,000 30 30
G726-40 sample 5 8,000 20
G726-32 sample 4 8,000 20
G726-24 sample 3 8,000 20
G726-16 sample 2 8,000 20
G728 frame N/A 8,000 2.5 20
G729 frame N/A 8,000 10 20
G729D frame N/A 8,000 10 20
G729E frame N/A 8,000 10 20
GSM frame N/A 8,000 20 20
GSM-EFR frame N/A 8,000 20 20
L8 sample 8 var. 20
L16 sample 16 var. 20
LPC frame N/A 8,000 20 20
MPA frame N/A var. var.
PCMA sample 8 var. 20
PCMU sample 8 var. 20
QCELP frame N/A 8,000 20 20
VDVI sample var. var. 20
Table 1: Properties of Audio Encodings (N/A: not applicable; var.:
The characteristics of the audio encodings described in this document
are shown in Table 1; they are listed in order of their payload type
in Table 4. While most audio codecs are only specified for a fixed
sampling rate, some sample-based algorithms (indicated by an entry of
"var." in the sampling rate column of Table 1) may be used with
different sampling rates, resulting in different coded bit rates.
When used with a sampling rate other than that for which a static
payload type is defined, non-RTP means beyond the scope of this memo
MUST be used to define a dynamic payload type and MUST indicate the
selected RTP timestamp clock rate, which is usually the same as the
sampling rate for audio.
DVI4 uses an adaptive delta pulse code modulation (ADPCM) encoding
scheme that was specified by the Interactive Multimedia Association
(IMA) as the "IMA ADPCM wave type". However, the encoding defined
here as DVI4 differs in three respects from the IMA specification:
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 specified a 3-bit-per-
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 IMA ADPCM algorithm was described in the document IMA Recommended
Practices for Enhancing Digital Audio Compatibility in Multimedia
Systems (version 3.0). However, the Interactive Multimedia
Association ceased operations in 1997. Resources for an archived
copy of that document and a software implementation of the RTP DVI4
encoding are listed in Section 13.
G722 is specified in ITU-T Recommendation G.722, "7 kHz audio-coding
within 64 kbit/s". The G.722 encoder produces a stream of octets,
each of which SHALL be octet-aligned in an RTP packet. The first bit
transmitted in the G.722 octet, which is the most significant bit of
the higher sub-band sample, SHALL correspond to the most significant
bit of the octet in the RTP packet.
Even though the actual sampling rate for G.722 audio is 16,000 Hz,
the RTP clock rate for the G722 payload format is 8,000 Hz because
that value was erroneously assigned in RFC 1890 and must remain
unchanged for backward compatibility. The octet rate or sample-pair
rate is 8,000 Hz.
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 a scalable channel coding scheme for wireless
applications 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
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. Receivers MUST accept both data rates and
MUST accept SID frames unless restriction of these capabilities has
been signaled. The MIME registration for G723 in RFC 3555 
specifies parameters that MAY be used with MIME or SDP to restrict to
a single data rate or to restrict the use of SID frames. 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 Rec. G.723.1 and is
the same as that produced by the G.723 C code reference
implementation. For the 6.3 kb/s data rate, this packing is
illustrated as follows, where the header (HDR) bits are always "0 0"
as shown in Fig. 1 to indicate operation at 6.3 kb/s, and the Z bit
is always set to zero. The diagrams show the bit packing in "network
byte order", also known as big-endian order. The bits of each 32-bit
word are numbered 0 to 31, with the most significant bit on the left
and numbered 0. The octets (bytes) of each word are transmitted most
significant octet first. The bits of each data field are numbered in
the order of the bit stream representation of the encoding (least
significant bit first). The vertical bars indicate the boundaries
between field fragments.
4.5.4 G726-40, G726-32, G726-24, and G726-16
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 8,000 samples/sec to and from a 40, 32, 24, or 16
kbit/s channel. The conversion is applied to the PCM stream using an
Adaptive Differential Pulse Code Modulation (ADPCM) transcoding
technique. The ADPCM representation consists of a series of
codewords with a one-to-one correspondence to the samples in the PCM
stream. The G726 data rates of 40, 32, 24, and 16 kbit/s have
codewords of 5, 4, 3, and 2 bits, respectively.
The 16 and 24 kbit/s encodings do not provide toll quality speech.
They are designed for used in overloaded Digital Circuit
Multiplication Equipment (DCME). ITU-T G.726 recommends that the 16
and 24 kbit/s encodings should be alternated with higher data rate
encodings to provide an average sample size of between 3.5 and 3.7
bits per sample.
The encodings of G.726 are here denoted as G726-40, G726-32, G726-24,
and G726-16. Prior to 1990, G721 described the 32 kbit/s ADPCM
encoding, and G723 described the 40, 32, and 16 kbit/s encodings.
Thus, G726-32 designates the same algorithm as G721 in RFC 1890.
A stream of G726 codewords contains no information on the encoding
being used, therefore transitions between G726 encoding types are not
permitted within a sequence of packed codewords. Applications MUST
determine the encoding type of packed codewords from the RTP payload
No payload-specific header information SHALL be included as part of
the audio data. A stream of G726 codewords MUST be packed into
octets as follows: the first codeword is placed into the first octet
such that the least significant bit of the codeword aligns with the
least significant bit in the octet, the second codeword is then
packed so that its least significant bit coincides with the least
significant unoccupied bit in the octet. When a complete codeword
cannot be placed into an octet, the bits overlapping the octet
boundary are placed into the least significant bits of the next
octet. Packing MUST end with a completely packed final octet. The
number of codewords packed will therefore be a multiple of 8, 2, 8,
and 4 for G726-40, G726-32, G726-24, and G726-16, respectively. An
example of the packing scheme for G726-32 codewords is as shown,
where bit 7 is the least significant bit of the first octet, and bit
A3 is the least significant bit of the first codeword:
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
|B B B B|A A A A|D D D D|C C C C| ...
|0 1 2 3|0 1 2 3|0 1 2 3|0 1 2 3|
An example of the packing scheme for G726-24 codewords follows, where
again bit 7 is the least significant bit of the first octet, and bit
A2 is the least significant bit of the first codeword:
0 1 2
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
|C C|B B B|A A A|F|E E E|D D D|C|H H H|G G G|F F| ...
|1 2|0 1 2|0 1 2|2|0 1 2|0 1 2|0|0 1 2|0 1 2|0 1|
Note that the "little-endian" direction in which samples are packed
into octets in the G726-16, -24, -32 and -40 payload formats
specified here is consistent with ITU-T Recommendation X.420, but is
the opposite of what is specified in ITU-T Recommendation I.366.2
Annex E for ATM AAL2 transport. A second set of RTP payload formats
matching the packetization of I.366.2 Annex E and identified by MIME
subtypes AAL2-G726-16, -24, -32 and -40 will be specified in a
G728 is specified in ITU-T Recommendation G.728, "Coding of speech at
16 kbit/s using low-delay code excited linear prediction".
A G.278 encoder translates 5 consecutive audio samples into a 10-bit
codebook index, resulting in a bit rate of 16 kb/s for audio sampled
at 8,000 samples per second. The group of five consecutive samples
is called a vector. Four consecutive vectors, labeled V1 to V4
(where V1 is to be played first by the receiver), build one G.728
frame. The four vectors of 40 bits are packed into 5 octets, labeled
B1 through B5. B1 SHALL be placed first in the RTP packet.
Referring to the figure below, the principle for bit order is
"maintenance of bit significance". Bits from an older vector are
more significant than bits from newer vectors. The MSB of the frame
goes to the MSB of B1 and the LSB of the frame goes to LSB of B5.
1 2 3 3
0 0 0 0 9
<------------- 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.
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. An implementation that signals or accepts
use of G729 payload format may implement either G.729 or G.729A
unless restricted by additional signaling specified elsewhere related
specifically to the encoding rather than the payload format. 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 . See the next Section
(4.5.7) for other data rates added in later G.729 Annexes. For all
data rates, the sampling frequency (and RTP timestamp clock rate) is
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.
Receivers MUST accept comfort noise frames if restriction of their
use has not been signaled. The MIME registration for G729 in RFC
3555  specifies a parameter that MAY be used with MIME or SDP to
restrict the use of comfort noise frames.
A G729 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 frames. The presence
of a comfort noise frame can be deduced from the length of the RTP
payload. The default packetization interval is 20 ms (two frames),
but in some situations it may be desirable to send 10 ms packets. An
example would be a transition from speech to comfort noise in the
first 10 ms of the packet. For some applications, a longer
packetization interval may be required to reduce the packet rate.
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|
| C1 | S1 | GA1 | GB1 | P2 | C2 |
| 1 1 1| | | | | |
|5 6 7 8 9 0 1 2|0 1 2 3|0 1 2|0 1 2 3|0 1 2 3 4|0 1 2 3 4 5 6 7|
| C2 | S2 | GA2 | GB2 |
| 1 1 1| | | |
|8 9 0 1 2|0 1 2 3|0 1 2|0 1 2 3|
Figure 4: G.729 and G.729A bit packing
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 in Fig. 4. The
diagrams show the bit packing in "network byte order", also known as
big-endian order. The bits of each 32-bit word are numbered 0 to 31,
with the most significant bit on the left and numbered 0. The octets
(bytes) of each word are transmitted most significant octet first.
The bits of each data field are numbered in the order as produced by
the G.729 C code reference implementation.
The packing of the G.729 Annex B comfort noise frame is shown in Fig.
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
|L| LSF1 | LSF2 | GAIN |R|
|S| | | |E|
|F| | | |S|
|0|0 1 2 3 4|0 1 2 3|0 1 2 3 4|V| RESV = Reserved (zero)
Figure 5: G.729 Annex B bit packing
4.5.7 G729D and G729E
Annexes D and E to ITU-T Recommendation G.729 provide additional data
rates. Because the data rate is not signaled in the bitstream, the
different data rates are given distinct RTP encoding names which are
mapped to distinct payload type numbers. G729D indicates a 6.4
kbit/s coding mode (G.729 Annex D, for momentary reduction in channel
capacity), while G729E indicates an 11.8 kbit/s mode (G.729 Annex E,
for improved performance with a wide range of narrow-band input
signals, e.g., music and background noise). Annex E has two
operating modes, backward adaptive and forward adaptive, which are
signaled by the first two bits in each frame (the most significant
two bits of the first octet).
The voice activity detector (VAD) and comfort noise generator (CNG)
algorithm specified in Annex B of G.729 may be used with Annex D and
Annex E frames in addition to G.729 and G.729 Annex A frames. The
algorithm details for the operation of Annexes D and E with the Annex
B CNG are specified in G.729 Annexes F and G. Note that Annexes F
and G do not introduce any new encodings. Receivers MUST accept
comfort noise frames if restriction of their use has not been
signaled. The MIME registrations for G729D and G729E in RFC 3555 
specify a parameter that MAY be used with MIME or SDP to restrict the
use of comfort noise frames.
For G729D, an RTP packet may consist of zero or more G.729 Annex D
frames, followed by zero or one G.729 Annex B frame. Similarly, for
G729E, an RTP packet may consist of zero or more G.729 Annex E
frames, followed by zero or one G.729 Annex B frame. The presence of
a comfort noise frame can be deduced from the length of the RTP
A single RTP packet must contain frames of only one data rate,
optionally followed by one comfort noise frame. The data rate may be
changed from packet to packet by changing the payload type number.
G.729 Annexes D, E and H describe what the encoding and decoding
algorithms must do to accommodate a change in data rate.
For G729D, the bits of a G.729 Annex D frame are formatted as shown
below in Fig. 6 (cf. Table D.1/G.729). The frame length is 64 bits.
22.214.171.124 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.
GSM-EFR denotes GSM 06.60 enhanced full rate speech transcoding,
specified in ETS 300 726 which is available from ETSI at the address
given in Section 4.5.8. 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
similar to that specified here for the original GSM 06.10 codec. The
packing is specified in ETSI Technical Specification TS 101 318.
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
L16 denotes uncompressed audio data samples, using 16-bit signed
representation with 65,535 equally divided steps between minimum and
maximum signal level, ranging from -32,768 to 32,767. The value is
represented in two's complement notation and transmitted in network
byte order (most significant byte first).
The MIME registration for L16 in RFC 3555  specifies parameters
that MAY be used with MIME or SDP to indicate that analog pre-
emphasis was applied to the signal before quantization or to indicate
that a multiple-channel audio stream follows a different channel
ordering convention than is specified in Section 4.1.
LPC designates an experimental linear predictive encoding contributed
by Ron Frederick, which is based on an implementation written by Ron
Zuckerman 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.
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 .
The encoding may be at any of three levels of complexity, called
Layer I, II and III. The selected layer as well as the sampling rate
and channel count are indicated in the payload. The RTP timestamp
clock rate is always 90,000, independent of the sampling rate.
MPEG-1 audio supports sampling rates of 32, 44.1, and 48 kHz (ISO/IEC
11172-3, section 1.1; "Scope"). MPEG-2 supports sampling rates of
16, 22.05 and 24 kHz. The number of samples per frame is fixed, but
the frame size will vary with the sampling rate and bit rate.
The MIME registration for MPA in RFC 3555  specifies parameters
that MAY be used with MIME or SDP to restrict the selection of layer,
channel count, sampling rate, and bit rate.
4.5.14 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 . 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 significant bit of the
octet as defined by the host machine format). The 56 kb/s and 48
kb/s modes of G.711 are not applicable to RTP, since PCMA and PCMU
MUST always be transmitted as 8-bit samples.
See Section 4.1 regarding silence suppression.
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 8,000 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 kb/s for normal mode and 4.7 kb/s
for reduced rate mode. The packetization of the QCELP audio codec is
described in .
The redundant audio payload format "RED" is specified by RFC 2198
. 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.
VDVI is a variable-rate version of DVI4, yielding speech bit rates of
between 10 and 25 kb/s. It is specified for single-channel operation
only. Samples are packed into octets starting at the most-
significant bit. The last octet is padded with 1 bits if the last
sample does not fill the last octet. This padding is distinct from
the valid codewords. The receiver needs to detect the padding
because there is no explicit count of samples in the packet.
It uses the following encoding:
DVI4 codeword VDVI bit pattern
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.
The CELL-B encoding is a proprietary encoding proposed by Sun
Microsystems. The byte stream format is described in RFC 2029 .
The encoding is specified in ISO Standards 10918-1 and 10918-2. The
RTP payload format is as specified in RFC 2435 .
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 .
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
. The H263-1998 payload format is RECOMMENDED over this one for
use by new implementations.
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
. 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.
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
, Section 3.
The MIME registration for MPV in RFC 3555  specifies a parameter
that MAY be used with MIME or SDP to restrict the selection of the
type of MPEG video.
MP2T designates the use of MPEG-2 transport streams, for either audio
or video. The RTP payload format is described in RFC 2250 ,
The encoding is implemented in the program `nv', version 4, developed
at Xerox PARC by Ron Frederick. Further information is available
from the author:
Blue Coat Systems Inc.
650 Almanor Avenue
Sunnyvale, CA 94085
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. Payload type 2 was assigned to G721 in
RFC 1890 and to its equivalent successor G726-32 in draft versions of
this specification, but its use is now deprecated and that static
payload type is marked reserved due to conflicting use for the
payload formats G726-32 and AAL2-G726-32 (see Section 4.5.4).
Payload type 13 indicates the Comfort Noise (CN) payload format
specified in RFC 3389 . Payload type 19 is marked "reserved"
because some draft versions of this specification assigned that
number to an earlier version of the comfort noise payload format.
The payload type range 72-76 is marked "reserved" 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. See the
section "Multiplexing RTP Sessions" of RFC 3550 for additional
PT encoding media type clock rate channels
0 PCMU A 8,000 1
1 reserved A
2 reserved A
3 GSM A 8,000 1
4 G723 A 8,000 1
5 DVI4 A 8,000 1
6 DVI4 A 16,000 1
7 LPC A 8,000 1
8 PCMA A 8,000 1
9 G722 A 8,000 1
10 L16 A 44,100 2
11 L16 A 44,100 1
12 QCELP A 8,000 1
13 CN A 8,000 1
14 MPA A 90,000 (see text)
15 G728 A 8,000 1
16 DVI4 A 11,025 1
17 DVI4 A 22,050 1
18 G729 A 8,000 1
19 reserved A
20 unassigned A
21 unassigned A
22 unassigned A
23 unassigned A
dyn G726-40 A 8,000 1
dyn G726-32 A 8,000 1
dyn G726-24 A 8,000 1
dyn G726-16 A 8,000 1
dyn G729D A 8,000 1
dyn G729E A 8,000 1
dyn GSM-EFR A 8,000 1
dyn L8 A var. var.
dyn RED A (see text)
dyn VDVI A var. 1
Table 4: Payload types (PT) for audio encodings
PT encoding media type clock rate
24 unassigned V
25 CelB V 90,000
26 JPEG V 90,000
27 unassigned V
28 nv V 90,000
29 unassigned V
30 unassigned V
31 H261 V 90,000
32 MPV V 90,000
33 MP2T AV 90,000
34 H263 V 90,000
35-71 unassigned ?
72-76 reserved N/A N/A
77-95 unassigned ?
96-127 dynamic ?
dyn H263-1998 V 90,000
Table 5: Payload types (PT) for video and combined
Session participants agree through mechanisms beyond the scope of
this specification on the set of payload types allowed in a given
session. This set MAY, for example, be defined by the capabilities
of the applications used, negotiated by a conference control protocol
or established by agreement between the human participants.
Audio applications operating under this profile SHOULD, at a minimum,
be able to send and/or receive payload types 0 (PCMU) and 5 (DVI4).
This allows interoperability without format negotiation and ensures
successful negotiation with a conference control protocol.
7. RTP over TCP and Similar Byte Stream Protocols
Under special circumstances, it may be necessary to carry RTP in
protocols offering a byte stream abstraction, such as TCP, possibly
multiplexed with other data. The application MUST define its own
method of delineating RTP and RTCP packets (RTSP  provides an
example of such an encapsulation specification).
8. Port Assignment
As specified in the RTP protocol definition, RTP data SHOULD be
carried on an even UDP 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 port
pair. For example, the port pair MAY be allocated randomly by a
session management program. A single fixed port number pair cannot
be required because multiple applications using this profile are
likely to run on the same host, and there are some operating systems
that do not allow multiple processes to use the same UDP port with
different multicast addresses.
However, port numbers 5004 and 5005 have been registered for use with
this profile for those applications that choose to use them as the
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 mostly backwards-compatible with
RFC 1890 except for functions removed because two interoperable
implementations were not found. The additions to RFC 1890 codify
existing practice in the use of payload formats under this profile.
Since this profile may be used without using any of the payload
formats listed here, the addition of new payload formats in this
revision does not affect backwards compatibility. The changes are
listed below, categorized into functional and non-functional changes.
o Section 11, "IANA Considerations" was added to specify the
registration of the name for this profile. That appendix also
references a new Section 3 "Registering Additional Encodings"
which establishes a policy that no additional registration of
static payload types for this profile will be made beyond those
added in this revision and included in Tables 4 and 5. Instead,
additional encoding names may be registered as MIME subtypes for
binding to dynamic payload types. Non-normative references were
added to RFC 3555  where MIME subtypes for all the listed
payload formats are registered, some with optional parameters for
use of the payload formats.
o Static payload types 4, 16, 17 and 34 were added to incorporate
IANA registrations made since the publication of RFC 1890, along
with the corresponding payload format descriptions for G723 and
o Following working group discussion, static payload types 12 and 18
were added along with the corresponding payload format
descriptions for QCELP and G729. Static payload type 13 was
assigned to the Comfort Noise (CN) payload format defined in RFC
3389. Payload type 19 was marked reserved because it had been
temporarily allocated to an earlier version of Comfort Noise
present in some draft revisions of this document.
o The payload format for G721 was renamed to G726-32 following the
ITU-T renumbering, and the payload format description for G726 was
expanded to include the -16, -24 and -40 data rates. Because of
confusion regarding draft revisions of this document, some
implementations of these G726 payload formats packed samples into
octets starting with the most significant bit rather than the
least significant bit as specified here. To partially resolve
this incompatibility, new payload formats named AAL2-G726-16, -24,
-32 and -40 will be specified in a separate document (see note in
Section 4.5.4), and use of static payload type 2 is deprecated as
explained in Section 6.
o Payload formats G729D and G729E were added following the ITU-T
addition of Annexes D and E to Recommendation G.729. Listings
were added for payload formats GSM-EFR, RED, and H263-1998
published in other documents subsequent to RFC 1890. These
additional payload formats are referenced only by dynamic payload
o The descriptions of the payload formats for G722, G728, GSM, VDVI
o The payload format for 1016 audio was removed and its static
payload type assignment 1 was marked "reserved" because two
interoperable implementations were not found.
o Requirements for congestion control were added in Section 2.
o This profile follows the suggestion in the revised RTP spec that
RTCP bandwidth may be specified separately from the session
bandwidth and separately for active senders and passive receivers.
o The mapping of a user pass-phrase string into an encryption key
was deleted from Section 2 because two interoperable
implementations were not found.
o The "quadrophonic" sample ordering convention for four-channel
audio was removed to eliminate an ambiguity as noted in Section
o In Section 4.1, it is now explicitly stated that silence
suppression is allowed for all audio payload formats. (This has
always been the case and derives from a fundamental aspect of
RTP's design and the motivations for packet audio, but was not
explicit stated before.) The use of comfort noise is also
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", and clarified that the marker bit is set only
when packets are intentionally not sent.
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 RFC references are added for payload formats published after RFC
o The security considerations and full copyright sections were
o According to Peter Hoddie of Apple, only pre-1994 Macintosh used
the 22254.54 rate and none the 11127.27 rate, so the latter was
dropped from the discussion of suggested sampling frequencies.
o Table 1 was corrected to move some values from the "ms/packet"
column to the "default ms/packet" column where they belonged.
o Since the Interactive Multimedia Association ceased operations, an
alternate resource was provided for a referenced IMA document.
o A note has been added for G722 to clarify a discrepancy between
the actual sampling rate and the RTP timestamp clock rate.
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
- A note was added that the convention for the order of channels
specified in Section 4.1 may be overridden by a particular
encoding or payload format specification.
- The terms MUST, SHOULD, MAY, etc. are used as defined in RFC
o A second author for this document was added.
10. Security Considerations
Implementations using the profile defined in this specification are
subject to the security considerations discussed in the RTP
specification . This profile does not specify any different
security services. 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
As with any IP-based protocol, in some circumstances a receiver may
be overloaded simply by the receipt of too many packets, either
desired or undesired. Network-layer authentication MAY be used to
discard packets from undesired sources, but the processing cost of
the authentication itself may be too high. In a multicast
environment, source pruning is implemented in IGMPv3 (RFC 3376) 
and in multicast routing protocols to allow a receiver to select
which sources are allowed to reach it.
11. IANA Considerations
The RTP specification establishes a registry of profile names for use
by higher-level control protocols, such as the Session Description
Protocol (SDP), RFC 2327 , to refer to transport methods. This
profile registers the name "RTP/AVP".
Section 3 establishes the policy that no additional registration of
static RTP payload types for this profile will be made beyond those
added in this document revision and included in Tables 4 and 5. IANA
may reference that section in declining to accept any additional
registration requests. In Tables 4 and 5, note that types 1 and 2
have been marked reserved and the set of "dyn" payload types included
has been updated. These changes are explained in Sections 6 and 9.
12.1 Normative References
 Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,
"RTP: A Transport Protocol for Real-Time Applications", RFC
3550, July 2003.
 Bradner, S., "Key Words for Use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
 Apple Computer, "Audio Interchange File Format AIFF-C", August
1991. (also ftp://ftp.sgi.com/sgi/aiff-c.9.26.91.ps.Z).
12.2 Informative References
 Braden, R., Clark, D. and S. Shenker, "Integrated Services in
the Internet Architecture: an Overview", RFC 1633, June 1994.
 Blake, S., Black, D., Carlson, M., Davies, E., Wang, Z. and W.
Weiss, "An Architecture for Differentiated Service", RFC 2475,
 Handley, M. and V. Jacobson, "SDP: Session Description
Protocol", RFC 2327, April 1998.
 Casner, S. and P. Hoschka, "MIME Type Registration of RTP
Payload Types", RFC 3555, July 2003.
 Freed, N., Klensin, J. and J. Postel, "Multipurpose Internet
Mail Extensions (MIME) Part Four: Registration Procedures", BCP
13, RFC 2048, November 1996.
 Zopf, R., "Real-time Transport Protocol (RTP) Payload for
Comfort Noise (CN)", RFC 3389, September 2002.
 Deleam, D. 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,
 Mouly, M. and M.-B. Pautet, The GSM system for mobile
communications Lassay-les-Chateaux, France: Europe Media
 Degener, J., "Digital Speech Compression", Dr. Dobb's Journal,
 Redl, S., Weber, M. and M. Oliphant, An Introduction to GSM
Boston: Artech House, 1995.
 Hoffman, D., Fernando, G., Goyal, V. and M. Civanlar, "RTP
Payload Format for MPEG1/MPEG2 Video", RFC 2250, January 1998.
 Jayant, N. and P. Noll, Digital Coding of Waveforms--Principles
and Applications to Speech and Video Englewood Cliffs, New
Jersey: Prentice-Hall, 1984.
 McKay, K., "RTP Payload Format for PureVoice(tm) Audio", RFC
2658, August 1999.
 Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., Handley, M.,
Bolot, J.-C., Vega-Garcia, A. and S. Fosse-Parisis, "RTP Payload
for Redundant Audio Data", RFC 2198, September 1997.
 Speer, M. and D. Hoffman, "RTP Payload Format of Sun's CellB
Video Encoding", RFC 2029, October 1996.
 Berc, L., Fenner, W., Frederick, R., McCanne, S. and P. Stewart,
"RTP Payload Format for JPEG-Compressed Video", RFC 2435,
 Turletti, T. and C. Huitema, "RTP Payload Format for H.261 Video
Streams", RFC 2032, October 1996.
 Zhu, C., "RTP Payload Format for H.263 Video Streams", RFC 2190,
 Bormann, C., Cline, L., Deisher, G., Gardos, T., Maciocco, C.,
Newell, D., Ott, J., Sullivan, G., Wenger, S. and C. Zhu, "RTP
Payload Format for the 1998 Version of ITU-T Rec. H.263 Video
(H.263+)", RFC 2429, October 1998.
 Schulzrinne, H., Rao, A. and R. Lanphier, "Real Time Streaming
Protocol (RTSP)", RFC 2326, April 1998.
 Cain, B., Deering, S., Kouvelas, I., Fenner, B. and A.
Thyagarajan, "Internet Group Management Protocol, Version 3",
RFC 3376, October 2002.
13. 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
An archived copy of the document IMA Recommended Practices for
Enhancing Digital Audio Compatibility in Multimedia Systems (version
3.0), which describes the IMA ADPCM algorithm, is available at:
An implementation is available from Jack Jansen at
An implementation of the G.722 algorithm is available as part of the
ITU-T STL, described above.
The reference C code implementation defining the G.723.1 algorithm
and its Annexes A, B, and C are available as an integral part of
Recommendation G.723.1 from the ITU Sales Service, address listed
above. Both the algorithm and C code are covered by a specific
license. The ITU-T Secretariat should be contacted to obtain such
G726 is specified in the ITU-T Recommendation G.726, "40, 32, 24, and
16 kb/s Adaptive Differential Pulse Code Modulation (ADPCM)". An
implementation of the G.726 algorithm is available as part of the
ITU-T STL, described above.
The reference C code implementation defining the G.729 algorithm and
its Annexes A through I are available as an integral part of
Recommendation G.729 from the ITU Sales Service, listed above. Annex
I contains the integrated C source code for all G.729 operating
modes. The G.729 algorithm and associated C code are covered by a
specific license. The contact information for obtaining the license
is available from the ITU-T Secretariat.
A reference implementation was written by Carsten Bormann and Jutta
Degener (then at TU Berlin, Germany). It is available at
Although the RPE-LTP algorithm is not an ITU-T standard, there is a C
code implementation of the RPE-LTP algorithm available as part of the
ITU-T STL. The STL implementation is an adaptation of the TU Berlin
An implementation is available at
An implementation of these algorithms is available as part of the
ITU-T STL, described above.
The comments and careful review of Simao Campos, Richard Cox and AVT
Working Group participants are gratefully acknowledged. The GSM
description was adopted from the IMTC Voice over IP Forum Service
Interoperability Implementation Agreement (January 1997). Fred Burg
and Terry Lyons helped with the G.729 description.
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16. Authors' Addresses
Department of Computer Science
1214 Amsterdam Avenue
New York, NY 10027
Stephen L. Casner
3400 Hillview Avenue, Building 3
Palo Alto, CA 94304
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