Internet Engineering Task Force AVT WG
Internet Draft Schulzrinne/Casner
draft-ietf-avt-profile-new-12.txt Columbia U./Packet Design
November 20, 2001
Expires: May 2002
RTP Profile for Audio and Video Conferences with Minimal Control
STATUS OF THIS MEMO
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all provisions of Section 10 of RFC2026.
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Abstract
This memorandum is a revision of RFC 1890 in preparation for
advancement from Proposed Standard to Draft Standard status.
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
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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.
Contents
1 Introduction ........................................ 3
1.1 Terminology ......................................... 3
2 RTP and RTCP Packet Forms and Protocol Behavior ..... 4
3 IANA Considerations ................................. 6
3.1 Registering Additional Encodings .................... 6
4 Audio ............................................... 8
4.1 Encoding-Independent Rules .......................... 8
4.2 Operating Recommendations ........................... 9
4.3 Guidelines for Sample-Based Audio Encodings ......... 10
4.4 Guidelines for Frame-Based Audio Encodings .......... 10
4.5 Audio Encodings ..................................... 11
4.5.1 DVI4 ................................................ 12
4.5.2 G722 ................................................ 13
4.5.3 G723 ................................................ 13
4.5.4 G726-40, G726-32, G726-24, and G726-16 .............. 17
4.5.5 G728 ................................................ 18
4.5.6 G729 ................................................ 19
4.5.7 G729D and G729E ..................................... 21
4.5.8 GSM ................................................. 24
4.5.8.1 General Packaging Issues ............................ 24
4.5.8.2 GSM variable names and numbers ...................... 25
4.5.9 GSM-EFR ............................................. 25
4.5.10 L8 .................................................. 25
4.5.11 L16 ................................................. 25
4.5.12 LPC ................................................. 27
4.5.13 MPA ................................................. 28
4.5.14 PCMA and PCMU ....................................... 28
4.5.15 QCELP ............................................... 28
4.5.16 RED ................................................. 28
4.5.17 VDVI ................................................ 29
5 Video ............................................... 29
5.1 CelB ................................................ 30
5.2 JPEG ................................................ 30
5.3 H261 ................................................ 30
5.4 H263 ................................................ 30
5.5 H263-1998 ........................................... 31
5.6 MPV ................................................. 31
5.7 MP2T ................................................ 31
5.8 nv .................................................. 31
6 Payload Type Definitions ............................ 31
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7 RTP over TCP and Similar Byte Stream Protocols ...... 32
8 Port Assignment ..................................... 32
9 Changes from RFC 1890 ............................... 34
10 Security Considerations ............................. 37
11 Full Copyright Statement ............................ 37
12 Acknowledgments ..................................... 38
13 Addresses of Authors ................................ 38
1 Introduction
[Note to the RFC Editor: This paragraph and the first paragraph of
the Abstract are to be deleted when this draft is published as an
RFC. 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. These latter references are intended to be non-
normative as this Profile may be used independently of the MIME
registrations.]
This profile defines aspects of RTP left unspecified in the RTP
Version 2 protocol definition (RFC XXXX) [2]. 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 3.
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. 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.
1.1 Terminology
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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 [1] 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.
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 [2] that 1/4 of the
RTCP bandwidth be dedicated to data senders, the
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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) [4] or Differentiated Services (RFC
2475) [5], 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 accordingly.
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
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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-layer protocols.)
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 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 [6], to refer to transport methods. This
profile registers the name "RTP/AVP".
3.1 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
RFCs.
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 [6], to identify encodings
selected for a particular RTP session.
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In some contexts it may be useful to refer to these encodings in the
form of a MIME content-type. To facilitate this, RFC YYYY [7]
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 [8].
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 YYYY 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, 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. 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
payload types.
Dynamic payload types SHOULD NOT be used without a well-defined
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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 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 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 usually
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 [3], using the following
notation, unless some other convention is specified for a particular
encoding or payload format:
l left
r right
c center
S surround
F front
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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.
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
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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
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
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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
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.:
variable)
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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.
4.5.1 DVI4
DVI4 is specified, with pseudo-code, in [9] 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 */
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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
Annapolis, MD 21401-8011
USA
phone: +1 410 626-1380
4.5.2 G722
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 16000 Hz, the
RTP clock rate for the G722 payload format is 8000 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
8000 Hz.
4.5.3 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
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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. The MIME registration for G723 in RFC YYYY
[7] 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.
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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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| LPC |HDR| LPC | LPC | ACL0 |LPC|
| | | | | | |
|0 0 0 0 0 0|0 0|1 1 1 1 0 0 0 0|2 2 1 1 1 1 1 1|0 0 0 0 0 0|2 2|
|5 4 3 2 1 0| |3 2 1 0 9 8 7 6|1 0 9 8 7 6 5 4|5 4 3 2 1 0|3 2|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ACL2 |ACL|A| GAIN0 |ACL|ACL| GAIN0 | GAIN1 |
| | 1 |C| | 3 | 2 | | |
|0 0 0 0 0|0 0|0|0 0 0 0|0 0|0 0|1 1 0 0 0 0 0 0|0 0 0 0 0 0 0 0|
|4 3 2 1 0|1 0|6|3 2 1 0|1 0|6 5|1 0 9 8 7 6 5 4|7 6 5 4 3 2 1 0|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| GAIN2 | GAIN1 | GAIN2 | GAIN3 | GRID | GAIN3 |
| | | | | | |
|0 0 0 0|1 1 0 0|1 1 0 0 0 0 0 0|0 0 0 0 0 0 0 0|0 0 0 0|1 1 0 0|
|3 2 1 0|1 0 9 8|1 0 9 8 7 6 5 4|7 6 5 4 3 2 1 0|3 2 1 0|1 0 9 8|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| MSBPOS |Z|POS| MSBPOS | POS0 |POS| POS0 |
| | | 0 | | | 1 | |
|0 0 0 0 0 0 0|0|0 0|1 1 1 0 0 0|0 0 0 0 0 0 0 0|0 0|1 1 1 1 1 1|
|6 5 4 3 2 1 0| |1 0|2 1 0 9 8 7|9 8 7 6 5 4 3 2|1 0|5 4 3 2 1 0|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| POS1 | POS2 | POS1 | POS2 | POS3 | POS2 |
| | | | | | |
|0 0 0 0 0 0 0 0|0 0 0 0|1 1 1 1|1 1 0 0 0 0 0 0|0 0 0 0|1 1 1 1|
|9 8 7 6 5 4 3 2|3 2 1 0|3 2 1 0|1 0 9 8 7 6 5 4|3 2 1 0|5 4 3 2|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| POS3 | PSIG0 |POS|PSIG2| PSIG1 | PSIG3 |PSIG2|
| | | 3 | | | | |
|1 1 0 0 0 0 0 0|0 0 0 0 0 0|1 1|0 0 0|0 0 0 0 0|0 0 0 0 0|0 0 0|
|1 0 9 8 7 6 5 4|5 4 3 2 1 0|3 2|2 1 0|4 3 2 1 0|4 3 2 1 0|5 4 3|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 1: G.723 (6.3 kb/s) bit packing
For the 5.3 kb/s data rate, the header (HDR) bits are always "0 1",
as shown in Fig. 2, to indicate operation at 5.3 kb/s.
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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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| LPC |HDR| LPC | LPC | ACL0 |LPC|
| | | | | | |
|0 0 0 0 0 0|0 1|1 1 1 1 0 0 0 0|2 2 1 1 1 1 1 1|0 0 0 0 0 0|2 2|
|5 4 3 2 1 0| |3 2 1 0 9 8 7 6|1 0 9 8 7 6 5 4|5 4 3 2 1 0|3 2|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ACL2 |ACL|A| GAIN0 |ACL|ACL| GAIN0 | GAIN1 |
| | 1 |C| | 3 | 2 | | |
|0 0 0 0 0|0 0|0|0 0 0 0|0 0|0 0|1 1 0 0 0 0 0 0|0 0 0 0 0 0 0 0|
|4 3 2 1 0|1 0|6|3 2 1 0|1 0|6 5|1 0 9 8 7 6 5 4|7 6 5 4 3 2 1 0|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| GAIN2 | GAIN1 | GAIN2 | GAIN3 | GRID | GAIN3 |
| | | | | | |
|0 0 0 0|1 1 0 0|1 1 0 0 0 0 0 0|0 0 0 0 0 0 0 0|0 0 0 0|1 1 0 0|
|3 2 1 0|1 0 9 8|1 0 9 8 7 6 5 4|7 6 5 4 3 2 1 0|4 3 2 1|1 0 9 8|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| POS0 | POS1 | POS0 | POS1 | POS2 |
| | | | | |
|0 0 0 0 0 0 0 0|0 0 0 0|1 1 0 0|1 1 0 0 0 0 0 0|0 0 0 0 0 0 0 0|
|7 6 5 4 3 2 1 0|3 2 1 0|1 0 9 8|1 0 9 8 7 6 5 4|7 6 5 4 3 2 1 0|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| POS3 | POS2 | POS3 | PSIG1 | PSIG0 | PSIG3 | PSIG2 |
| | | | | | | |
|0 0 0 0|1 1 0 0|1 1 0 0 0 0 0 0|0 0 0 0|0 0 0 0|0 0 0 0|0 0 0 0|
|3 2 1 0|1 0 9 8|1 0 9 8 7 6 5 4|3 2 1 0|3 2 1 0|3 2 1 0|3 2 1 0|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 2: G.723 (5.3 kb/s) bit packing
The packing of G.723.1 SID (silence) frames, which are indicated by
the header (HDR) bits having the pattern "1 0", is depicted in Fig.
3.
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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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| LPC |HDR| LPC | LPC | GAIN |LPC|
| | | | | | |
|0 0 0 0 0 0|1 0|1 1 1 1 0 0 0 0|2 2 1 1 1 1 1 1|0 0 0 0 0 0|2 2|
|5 4 3 2 1 0| |3 2 1 0 9 8 7 6|1 0 9 8 7 6 5 4|5 4 3 2 1 0|3 2|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 3: G.723 SID mode bit packing
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 8000 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 is not
permitted within a sequence of packed codewords. Applications MUST
determine the encoding type of packed codewords from the RTP payload
identifier.
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
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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:
0 1
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 is:
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|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
4.5.5 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.
4.5.6 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 [10]. 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 8000 Hz.
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
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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. The
MIME registration for G729 in RFC YYYY [7] 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.
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.
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 packing of the G.729 Annex B comfort noise frame is shown in Fig.
5.
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0 1
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. The MIME registrations for
G729D and G729E in RFC YYYY [7] 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
payload.
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
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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.
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 | C1 |
|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 5|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| C1 |S1 | GA1 | GB1 | P2 | C2 |S2 | GA2 | GB2 |
| | | | | | | | | |
|6 7 8|0 1|0 1 2|0 1 2|0 1 2 3|0 1 2 3 4 5 6 7 8|0 1|0 1 2|0 1 2|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 6: G.729 Annex D bit packing
The net bit rate for the G.729 Annex E algorithm is 11.8 kbit/s and a
total of 118 bits are used. Two bits are appended as "don't care"
bits to complete an integer number of octets for the frame. For
G729E, the bits of a data frame are formatted as shown in the next
two diagrams (cf. Table E.1/G.729). The fields for the G729E forward
adaptive mode are packed as shown in Fig. 7.
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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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0 0|L| L1 | L2 | L3 | P1 |P| C0_1|
| |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|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| | C1_1 | C2_1 | C3_1 | C4_1 |
| | | | | |
|3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| GA1 | GB1 | P2 | C0_2 | C1_2 | C2_2 |
| | | | | | |
|0 1 2|0 1 2 3|0 1 2 3 4|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| | C3_2 | C4_2 | GA2 | GB2 |DC |
| | | | | | |
|6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2|0 1 2 3|0 1|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 7: G.729 Annex E (forward adaptive mode) bit packing
The fields for the G729E backward adaptive mode are packed as shown
in Fig. 8.
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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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1 1| P1 |P| C0_1 | C1_1 |
| | |0| 1 1 1| |
| |0 1 2 3 4 5 6 7|0|0 1 2 3 4 5 6 7 8 9 0 1 2|0 1 2 3 4 5 6 7|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| | C2_1 | C3_1 | C4_1 |GA1 | GB1 |P2 |
| | | | | | | |
|8 9|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2|0 1 2 3|0 1|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| | C0_2 | C1_2 | C2_2 |
| | 1 1 1| | |
|2 3 4|0 1 2 3 4 5 6 7 8 9 0 1 2|0 1 2 3 4 5 6 7 8 9|0 1 2 3 4 5|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| | C3_2 | C4_2 | GA2 | GB2 |DC |
| | | | | | |
|6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2|0 1 2 3|0 1|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 8: G.729 Annex E (backward adaptive mode) bit packing
4.5.8 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 [11,12,13]. 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.
4.5.8.1 General Packaging Issues
The GSM standard (ETS 300 961) specifies the bit stream produced by
the codec, but does not specify how these bits should be packed for
transmission. The packetization specified here has subsequently been
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adopted in ETSI Technical Specification TS 101 318. 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.
4.5.8.2 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.
4.5.9 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 4.5.8. 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.
4.5.10 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.
4.5.11 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).
The MIME registration for L16 in RFC YYYY [7] specifies parameters
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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
that MAY be used with MIME or SDP to indicate that analog preemphasis
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.
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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
4.5.12 LPC
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.
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4.5.13 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 [14].
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 90000, 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 YYYY [7] 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 [15]. 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.
4.5.15 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 [16].
4.5.16 RED
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The redundant audio payload format "RED" is specified by RFC 2198
[17]. 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.
4.5.17 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. 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
_______________________________
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
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.
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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 CelB
The CELL-B encoding is a proprietary encoding proposed by Sun
Microsystems. The byte stream format is described in RFC 2029 [18].
5.2 JPEG
The encoding is specified in ISO Standards 10918-1 and 10918-2. The
RTP payload format is as specified in RFC 2435 [19].
5.3 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 [20].
5.4 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
[21]. The H263-1998 payload format is RECOMMENDED over this one for
use by new implementations.
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5.5 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
[22]. 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.
5.6 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
[14], Section 3.
The MIME registration for MPV in RFC YYYY [7] specifies a parameter
that MAY be used with MIME or SDP to restrict the selection of the
type of MPEG video.
5.7 MP2T
MP2T designates the use of MPEG-2 transport streams, for either audio
or video. The RTP payload format is described in RFC 2250 [14],
Section 2.
5.8 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
Cacheflow Inc.
650 Almanor Avenue
Sunnyvale, CA 94085
United States
electronic mail: ronf@cacheflow.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
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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 13 is reserved for a
comfort noise payload format to be specified in a separate RFC.
Payload type 19 is also marked "reserved" because some draft versions
of this specification assigned that number to a 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 XXXX for additional
explanation.
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 [23] provides an
example of such an encapsulation specification.)
8 Port Assignment
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PT encoding media type clock rate channels
name (Hz)
___________________________________________________
0 PCMU A 8000 1
1 reserved A
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 8000 1
10 L16 A 44100 2
11 L16 A 44100 1
12 QCELP A 8000 1
13 reserved 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 reserved A
20 unassigned A
21 unassigned A
22 unassigned A
23 unassigned A
dyn G726-40 A 8000 1
dyn G726-24 A 8000 1
dyn G726-16 A 8000 1
dyn G729D A 8000 1
dyn G729E A 8000 1
dyn GSM-EFR A 8000 1
dyn L8 A var. var.
dyn RED A (see text)
dyn VDVI A var. 1
Table 4: Payload types (PT) for audio encodings
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
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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 H263-1998 V 90000
Table 5: Payload types (PT) for video and combined encodings
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
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revision does not affect backwards compatibility. The changes are
listed below, categorized into functional and non-functional changes.
Functional changes:
o A new Section "IANA Considerations" was added to specify the
registration of the name for this profile and to establish a
new 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 YYYY [7] 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 H263.
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
reserved for a comfort noise payload format to be defined in a
separate RFC. Payload type 19 was marked reserved because it
had been temporarily allocated in some draft revisions of this
document.
o The payload format for G721 was renamed to G726-32 following
the ITU-T renumbering.
o The payload format description for G726 was expanded to
include the -16, -24 and -40 data rates. 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 type numbers.
o The descriptions of the payload formats for G722, G728, GSM,
VDVI were expanded.
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.
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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.
Non-functional changes:
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 1890.
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 Table 1 was corrected to move some values from the "ms/packet"
column to the "default ms/packet" column where they belonged.
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.
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- 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.
- 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
2119.
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 [2]. 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
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
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 [24] 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 (2001). All Rights Reserved.
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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 implementation 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 Acknowledgments
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.
13 Addresses of Authors
Henning Schulzrinne
Dept. of Computer Science
Columbia University
1214 Amsterdam Avenue
New York, NY 10027
USA
electronic mail: schulzrinne@cs.columbia.edu
Stephen L. Casner
Packet Design
2465 Latham Street
Mountain View, CA 94040
United States
electronic mail: casner@acm.org
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References
Normative References
[1] S. Bradner, "Key words for use in RFCs to Indicate Requirement
Levels," RFC 2119, Internet Engineering Task Force, Mar. 1997.
[2] 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.
[3] Apple Computer, "Audio interchange file format AIFF-C," Aug.
1991. (also ftp://ftp.sgi.com/sgi/aiff-c.9.26.91.ps.Z).
Non-Normative References
[4] R. Braden, D. Clark, S. Shenker, "Integrated Services in the
Internet Architecture: an Overview," Request for Comments
(Informational) RFC 1633, Internet Engineering Task Force, June 1994.
[5] S. Blake, D. Black, M. Carlson, E. Davies, Z. Wang, W. Weiss, "An
Architecture for Differentiated Service," Request for Comments
(Proposed Standard) RFC 2475, Internet Engineering Task Force, Dec.
1998.
[6] M. Handley and V. Jacobson, "SDP: Session Description Protocol,"
Request for Comments (Proposed Standard) RFC 2327, Internet
Engineering Task Force, Apr. 1998.
[7] S. Casner and P. Hoschka, "MIME Type Registration of RTP Payload
Types," Internet Draft, Internet Engineering Task Force, July 2001.
Work in progress.
[8] N. Freed, J. Klensin, and J. Postel, "Multipurpose Internet Mail
Extensions (MIME) Part Four: Registration Procedures," RFC 2048,
Internet Engineering Task Force, Nov. 1996.
[9] 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.
[10] 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
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Conference on Signal Processing, Technology, and Applications
(ICSPAT) , (Boston, Massachusetts), pp. 1656--1660, Oct. 1996.
[11] M. Mouly and M.-B. Pautet, The GSM system for mobile
communications Lassay-les-Chateaux, France: Europe Media Duplication,
1993.
[12] J. Degener, "Digital speech compression," Dr. Dobb's Journal ,
Dec. 1994.
[13] S. M. Redl, M. K. Weber, and M. W. Oliphant, An Introduction to
GSM Boston: Artech House, 1995.
[14] 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.
[15] N. S. Jayant and P. Noll, Digital Coding of Waveforms--
Principles and Applications to Speech and Video Englewood Cliffs, New
Jersey: Prentice-Hall, 1984.
[16] K. McKay, "RTP Payload Format for PureVoice(tm) Audio", Request
for Comments (Proposed Standard) RFC 2658, Internet Engineering Task
Force, Aug. 1999.
[17] 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.
[18] 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.
[19] 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.
[20] 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.
[21] C. Zhu, "RTP payload format for H.263 video streams," Request
for Comments (Proposed Standard) RFC 2190, Internet Engineering Task
Force, Sep. 1997.
[22] C. Bormann, L. Cline, G. Deisher, T. Gardos, C. Maciocco, D.
Newell, J. Ott, G. Sullivan, S. Wenger, C. Zhu, "RTP Payload Format
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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.
[23] 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.
[24] 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
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
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license. The ITU-T Secretariat should be contacted to obtain such
licensing information.
G726
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.
G729
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.
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 [9].
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