draft-ietf-avt-profile-new-00.txt   draft-ietf-avt-profile-new-01.txt 
Internet Engineering Task Force AVT WG Internet Engineering Task Force AVT WG
Internet Draft Schulzrinne Internet Draft Schulzrinne
ietf-avt-profile-new-00.txt Columbia U. ietf-avt-profile-new-01.txt Columbia U.
March 26, 1997 July 29, 1997
Expires: September 9, 1997 Expires: January 1, 1998
RTP Profile for Audio and Video Conferences with Minimal Control RTP Profile for Audio and Video Conferences with Minimal Control
STATUS OF THIS MEMO STATUS OF THIS MEMO
This document is an Internet-Draft. Internet-Drafts are working This document is an Internet-Draft. Internet-Drafts are working
documents of the Internet Engineering Task Force (IETF), its areas, documents of the Internet Engineering Task Force (IETF), its areas,
and its working groups. Note that other groups may also distribute and its working groups. Note that other groups may also distribute
working documents as Internet-Drafts. working documents as Internet-Drafts.
skipping to change at page 2, line 16 skipping to change at page 2, line 16
provide pointers to reference implementations and the provide pointers to reference implementations and the
detailed standards. This document is meant as an aid for detailed standards. This document is meant as an aid for
implementors of audio, video and other real-time implementors of audio, video and other real-time
multimedia applications. multimedia applications.
Changes Changes
This draft revises RFC 1890. It is fully backwards-compatible with This draft revises RFC 1890. It is fully backwards-compatible with
RFC 1890 and codifies existing practice. It is intended that this RFC 1890 and codifies existing practice. It is intended that this
draft form the basis of a new RFC to obsolete RFC 1890 as it moves to draft form the basis of a new RFC to obsolete RFC 1890 as it moves to
Draft Standard.. Draft Standard.
Besides wording clarifications and filling in RFC numbers for payload Besides wording clarifications and filling in RFC numbers for payload
type definitions, this draft adds payload types 4, 13, 16, 17, 18 and type definitions, this draft adds payload types 4, 16, 17, 18, 19 and
34. The PostScript version of this draft contains change bars. 34. The PostScript version of this draft contains change bars marking
changes make since draft -00.
A tentative TCP encapsulation is defined.
According to Peter Hoddie of Apple, only pre-1994 Macintosh used the
22254.54 rate and none the 11127.27 rate.
Note to RFC editor: This section is to be removed before publication Note to RFC editor: This section is to be removed before publication
as an RFC. All RFC TBD should be filled in with the number of the RTP as an RFC. All RFC TBD should be filled in with the number of the RTP
specification RFC submitted for DS status. specification RFC submitted for Draft Standard status.
1 Introduction 1 Introduction
This profile defines aspects of RTP left unspecified in the RTP This profile defines aspects of RTP left unspecified in the RTP
Version 2 protocol definition (RFC XXXX). This profile is intended Version 2 protocol definition (RFC XXXX). This profile is intended
for the use within audio and video conferences with minimal session for the use within audio and video conferences with minimal session
control. In particular, no support for the negotiation of parameters control. In particular, no support for the negotiation of parameters
or membership control is provided. The profile is expected to be or membership control is provided. The profile is expected to be
useful in sessions where no negotiation or membership control are useful in sessions where no negotiation or membership control are
used (e.g., using the static payload types and the membership used (e.g., using the static payload types and the membership
skipping to change at page 4, line 4 skipping to change at page 4, line 10
within that slot and the remaining SDES items cyclically taking within that slot and the remaining SDES items cyclically taking
up the eighth slot, as defined in Section 6.2.2 of the RTP 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, 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 4, 7, 10, 13, 16, 19, while, say, EMAIL is used in RTCP packet
22. 22.
Security: The RTP default security services are also the default Security: The RTP default security services are also the default
under this profile. under this profile.
String-to-key mapping: A user-provided string ("pass phrase") is String-to-key mapping: A user-provided string ("pass phrase") is
hashed with the MD5 algorithm to a 16-octet digest. An hashed with the MD5 algorithm to a 16-octet digest. An !n!-bit
key is extracted from the digest by taking the first !n! bits
-bit key is extracted from the digest by taking the first from the digest. If several keys are needed with a total length
of 128 bits or less (as for triple DES), they are extracted in
bits from the digest. If several keys are needed with a total order from that digest. The octet ordering is specified in RFC
length of 128 bits or less (as for triple DES), they are 1423, Section 2.2. (Note that some DES implementations require
extracted in order from that digest. The octet ordering is that the 56-bit key be expanded into 8 octets by inserting an
specified in RFC 1423, Section 2.2. (Note that some DES odd parity bit in the most significant bit of the octet to go
implementations require that the 56-bit key be expanded into 8 with each 7 bits of the key.)
octets by inserting an odd parity bit in the most significant
bit of the octet to go with each 7 bits of the key.)
It is suggested that pass phrases are restricted to ASCII letters, It is suggested that pass phrases are restricted to ASCII letters,
digits, the hyphen, and white space to reduce the the chance of digits, the hyphen, and white space to reduce the the chance of
transcription errors when conveying keys by phone, fax, telex or transcription errors when conveying keys by phone, fax, telex or
email. email.
The pass phrase may be preceded by a specification of the encryption The pass phrase may be preceded by a specification of the encryption
algorithm. Any characters up to the first slash (ASCII 0x2f) are algorithm. Any characters up to the first slash (ASCII 0x2f) are
taken as the name of the encryption algorithm. The encryption format taken as the name of the encryption algorithm. The encryption format
specifiers should be drawn from RFC 1423 or any additional specifiers should be drawn from RFC 1423 or any additional
skipping to change at page 4, line 46 skipping to change at page 4, line 50
8859-1 characters do); (2) remove leading and trailing white space 8859-1 characters do); (2) remove leading and trailing white space
characters; (3) replace one or more contiguous white space characters characters; (3) replace one or more contiguous white space characters
by a single space (ASCII or UTF-8 0x20); (4) convert all letters to by a single space (ASCII or UTF-8 0x20); (4) convert all letters to
lower case and replace sequences of characters and non-spacing lower case and replace sequences of characters and non-spacing
accents with a single character, where possible. A minimum length of accents with a single character, where possible. A minimum length of
16 key characters (after applying the transformation) should be 16 key characters (after applying the transformation) should be
enforced by the application, while applications must allow up to 256 enforced by the application, while applications must allow up to 256
characters of input. characters of input.
Underlying protocol: The profile specifies the use of RTP over Underlying protocol: The profile specifies the use of RTP over
unicast and multicast UDP. (This does not preclude the use of unicast and multicast UDP as well as TCP. (This does not
these definitions when RTP is carried by other lower-layer preclude the use of these definitions when RTP is carried by
protocols.) other lower-layer protocols.)
Transport mapping: The standard mapping of RTP and RTCP to Transport mapping: The standard mapping of RTP and RTCP to
transport-level addresses is used. transport-level addresses is used.
Encapsulation: No encapsulation of RTP packets is specified. Encapsulation: No encapsulation of RTP packets is specified.
3 Registering Payload Types 3 Registering Payload Types
This profile defines a set of standard encodings and their payload This profile defines a set of standard encodings and their payload
types when used within RTP. Other encodings and their payload types types when used within RTP. Other encodings and their payload types
are to be registered with the Internet Assigned Numbers Authority are to be registered with the Internet Assigned Numbers Authority
(IANA). When registering a new encoding/payload type, the following (IANA). When registering a new encoding/payload type, the following
information should be provided: information should be provided:
o name and description of encoding, in particular the RTP o name and description of encoding, in particular the RTP
timestamp clock rate; the names defined here are 3 or 4 timestamp clock rate; the names defined here are 3 or 4
characters long to allow a compact representation if needed; characters long to allow a compact representation if needed;
o indication of who has change control over the encoding (for o indication of who has change control over the encoding (for
example, ISO, CCITT/ITU, other international standardization example, ISO, ITU-T, other international standardization
bodies, a consortium or a particular company or group of bodies, a consortium or a particular company or group of
companies); companies);
o any operating parameters or profiles; o any operating parameters or profiles;
o a reference to a further description, if available, for o a reference to a further description, if available, for
example (in order of preference) an RFC, a published paper, a example (in order of preference) an RFC, a published paper, a
patent filing, a technical report, documented source code or a patent filing, a technical report, documented source code or a
computer manual; computer manual;
o for proprietary encodings, contact information (postal and o for proprietary encodings, contact information (postal and
email address); email address);
o the payload type value for this profile, if necessary (see o the payload type value for this profile, if necessary (see
below). below).
Note that not all encodings to be used by RTP need to be assigned a Note that not all encodings to be used by RTP need to be assigned a
static payload type. Non-RTP means beyond the scope of this memo static payload type. Non-RTP means beyond the scope of this memo
(such as directory services or invitation protocols) may be used to (such as directory services or invitation protocols) may be used to
establish a dynamic mapping between a payload type drawn from the establish a dynamic mapping between a payload type and an encoding
range ("dynamic payload types"). Applications should first use the range 96
to 127 for dynamic payload types. Only applications which need to
and an encoding. For implementor convenience, this profile contains define more than 32 dynamic payload types may redefine codes below
descriptions of encodings which do not currently have a static 96. Redefining payload types below 96 may cause incorrect operation
payload type assigned to them. if an attempt is made to join a session without obtaining session
description information that defines the dynamic payload types.
Note that dynamic payload types should not be used without a well- Note that dynamic payload types should not be used without a well-
defined mechanism to indicate the mapping. Systems that expect to defined mechanism to indicate the mapping. Systems that expect to
interoperate with others operating under this profile should not interoperate with others operating under this profile should not
assign proprietary encodings to particular, fixed payload types in assign proprietary encodings to particular, fixed payload types in
the range reserved for dynamic payload types. the range reserved for dynamic payload types. SDP (RFC XXXX ) defines
such a mapping mechanism.
The available payload type space is relatively small. Thus, new The available payload type space is relatively small. Thus, new
static payload types are assigned only if the following conditions static payload types are assigned only if the following conditions
are met: are met:
o The encoding is of interest to the Internet community at o The encoding is of interest to the Internet community at
large. large.
o It offers benefits compared to existing encodings and/or is o It offers benefits compared to existing encodings and/or is
required for interoperation with existing, widely deployed required for interoperation with existing, widely deployed
conferencing or multimedia systems. conferencing or multimedia systems.
o The description is sufficient to build a decoder. o The description is sufficient to build a decoder.
The four-character encoding names are those those by the Session For implementor convenience, this profile contains descriptions of
Description Protocol (SDP) (RFC XXXX) . encodings which do not currently have a static payload type assigned
to them.
The Session Description Protocol (SDP) (RFC XXXX) uses the encoding
names defined here.
4 Audio 4 Audio
4.1 Encoding-Independent Rules 4.1 Encoding-Independent Rules
For applications which send no packets during silence, the first For applications which send no packets during silence, the first
packet of a talkspurt, that is, the first packet after a silence packet of a talkspurt, that is, the first packet after a silence
period, is distinguished by setting the marker bit in the RTP data period, is distinguished by setting the marker bit in the RTP data
header. The beginning of a talkspurt may be used to adjust the header to one. The marker bits in all other packets is zero. The
playout delay to reflect changing network delays. Applications beginning of a talkspurt may be used to adjust the playout delay to
without silence suppression set the bit to zero. reflect changing network delays. Applications without silence
suppression set the bit to zero.
The RTP clock rate used for generating the RTP timestamp is The RTP clock rate used for generating the RTP timestamp is
independent of the number of channels and the encoding; it equals the independent of the number of channels and the encoding; it equals the
number of sampling periods per second. For number of sampling periods per second. For !N!-channel encodings,
each sampling period (say, 1/8000 of a second) generates !N! samples.
-channel encodings, each sampling period (say, (This terminology is standard, but somewhat confusing, as the total
number of samples generated per second is then the sampling rate
of a second) generates times the channel count.)
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- If multiple audio channels are used, channels are numbered left-to-
right, starting at one. In RTP audio packets, information from right, starting at one. In RTP audio packets, information from
lower-numbered channels precedes that from higher-numbered channels. lower-numbered channels precedes that from higher-numbered channels.
For more than two channels, the convention followed by the AIFF-C For more than two channels, the convention followed by the AIFF-C
audio interchange format should be followed [1], using the following audio interchange format should be followed [1], using the following
notation: notation:
l left l left
r right r right
c center c center
S surround S surround
F front F front
R rear R rear
skipping to change at page 7, line 32 skipping to change at page 7, line 33
4 l c r S 4 l c r S
5 Fl Fr Fc Sl Sr 5 Fl Fr Fc Sl Sr
6 l lc c r rc S 6 l lc c r rc S
Samples for all channels belonging to a single sampling instant must Samples for all channels belonging to a single sampling instant must
be within the same packet. The interleaving of samples from different be within the same packet. The interleaving of samples from different
channels depends on the encoding. General guidelines are given in channels depends on the encoding. General guidelines are given in
Section 4.3 and 4.4. Section 4.3 and 4.4.
The sampling frequency should be drawn from the set: 8000, 11025, The sampling frequency should be drawn from the set: 8000, 11025,
16000, 22050, 24000, 32000, 44100 and 48000 Hz. (The Apple Macintosh 16000, 22050, 24000, 32000, 44100 and 48000 Hz. (Older Apple
computers have native sample rates of 22254.54 and 11127.27, which Macintosh computers had a native sample rate of 22254.54 Hz, which
can be converted to 22050 and 11025 with acceptable quality by can be converted to 22050 with acceptable quality by dropping 4
dropping 4 or 2 samples in a 20 ms frame.) However, most audio samples in a 20 ms frame.) However, most audio encodings are defined
encodings are defined for a more restricted set of sampling for a more restricted set of sampling frequencies. Receivers should
frequencies. Receivers should be prepared to accept multi-channel be prepared to accept multi-channel audio, but may choose to only
audio, but may choose to only play a single channel. play a single channel.
4.2 Operating Recommendations 4.2 Operating Recommendations
The following recommendations are default operating parameters. The following recommendations are default operating parameters.
Applications should be prepared to handle other values. The ranges Applications should be prepared to handle other values. The ranges
given are meant to give guidance to application writers, allowing a given are meant to give guidance to application writers, allowing a
set of applications conforming to these guidelines to interoperate set of applications conforming to these guidelines to interoperate
without additional negotiation. These guidelines are not intended to without additional negotiation. These guidelines are not intended to
restrict operating parameters for applications that can negotiate a restrict operating parameters for applications that can negotiate a
set of interoperable parameters, e.g., through a conference control set of interoperable parameters, e.g., through a conference control
skipping to change at page 9, line 25 skipping to change at page 9, line 28
number of frames per packet. number of frames per packet.
RTP packets shall contain a whole number of frames, with frames RTP packets shall contain a whole number of frames, with frames
inserted according to age within a packet, so that the oldest frame inserted according to age within a packet, so that the oldest frame
(to be played first) occurs immediately after the RTP packet header. (to be played first) occurs immediately after the RTP packet header.
The RTP timestamp reflects the capturing time of the first sample in The RTP timestamp reflects the capturing time of the first sample in
the first frame, that is, the oldest information in the packet. the first frame, that is, the oldest information in the packet.
4.5 Audio Encodings 4.5 Audio Encodings
encoding sample/frame bits/sample ms/frame ms/packet The characteristics of standard audio encodings are shown in Table 1;
________________________________________________________________ those assigned static payload types are listed in Table 3. While most
1016 frame N/A 30 30 audio codecs are only specified for a fixed sampling rate, some
DVI4 sample 4 20 sample-based algorithms (indicated by an entry of "var." in the
G721 sample 4 20 sampling rate column of Table 1) may be used with different sampling
G722 sample 8 20 rates, resulting in different coded bit rates. Non-RTP means MUST
G723 frame N/A 30 30 indicate the appropriate sampling rate.
G728 frame N/A 2.5 20
G729 frame N/A 10 20
GSM frame N/A 20 20
L8 sample 8 20
L16 sample 16 20
LPC frame N/A 20 20
MPA frame N/A 20
PCMA sample 8 20
PCMU sample 8 20
VDVI sample var. 20
Table 1: Properties of Audio Encodings
The characteristics of standard audio encodings are shown in Table 1
and their payload types are listed in Table 4.
4.5.1 1016 4.5.1 1016
Encoding 1016 is a frame based encoding using code-excited linear Encoding 1016 is a frame based encoding using code-excited linear
prediction (CELP) and is specified in Federal Standard FED-STD 1016 prediction (CELP) and is specified in Federal Standard FED-STD 1016
[2,3,4,5]. [2,3,4,5].
The U. S. DoD's Federal-Standard-1016 based 4800 bps code excited
linear prediction voice coder version 3.2 (CELP 3.2) Fortran and C
simulation source codes are available for worldwide distribution at
no charge (on DOS diskettes, but configured to compile on Sun SPARC
stations) from: Bob Fenichel, National Communications System,
Washington, D.C. 20305, phone +1-703-692-2124, fax +1-703-746-4960.
4.5.2 CN 4.5.2 CN
The G.764-based VAD (voice activity detector) noise level packet The CN (comfort noise) packet contains a single-octet message to the
contains a single-octet message to the receiver to play comfort noise receiver to play comfort noise at the absolute level specified. This
at the absolute dBmO level specified by the G.764 level index. This
message would normally be sent once at the beginning of a silence message would normally be sent once at the beginning of a silence
period (which also indicates the transition from speech to silence), period (which also indicates the transition from speech to silence),
but rate of noise level updates is implementation specific. The but rate of noise level updates is implementation specific. The
mapping of the index to absolute noise levels measured on the magnitude of the noise level is packed into the least significant
transmit side is given in Table 2, with the level index packed into bits of the noise-level payload, as shown below.
the least significant bits of the noise-level payload, as shown
below.
0 name of sampling default
encoding sample/frame bits/sample rate ms/frame ms/packet
____________________________________________________________________________
1016 frame N/A 8,000 30 30
CN frame N/A var.
DVI4 sample 4 var. 20
G722 sample 8 16,000 20
G723 frame N/A 8,000 30 30
G726-16 sample 2 8,000 20
G726-24 sample 3 8,000 20
G726-32 sample 4 8,000 20
G726-40 sample 5 8,000 20
G727-16 sample 2 8,000 20
G727-24 sample 3 8,000 20
G727-32 sample 4 8,000 20
G727-40 sample 5 8,000 20
G728 frame N/A 8,000 2.5 20
G729 frame N/A 8,000 10 20
GSM frame N/A 8,000 20 20
L8 sample 8 var. 20
L16 sample 16 var. 20
LPC frame N/A 8,000 20 20
MPA frame N/A var. 20
PCMA sample 8 var. 20
PCMU sample 8 var. 20
SX7300P frame N/A 8,000 15 30
SX8300P frame N/A 8,000 15 30
VDVI sample var. var. 20
Table 1: Properties of Audio Encodings (N/A: not applicable; var.:
variable)
The noise level is expressed in dBov, with values from 0 to 63 dBov.
dBov is the level relative to the overload of the system. (Note:
Representation relative to the overload point of a system is
particularly useful for digital implementations, since one does not
need to know the relative calibration of the analog circuitry.)
Example: In 16-bit linear PCM system (L16), a signal with 0 dBov
represents a square wave with the maximum possible amplitude (+/-
32767). -63 dBov corresponds to -58 dBm0 in a standard telephone
system. (dBm is the power level in decibels relative to 1 mW, with an
impedance of 600 Ohms.)
0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+
|0 0 0 0| level | |0 0| level |
+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+
The RTP header for the comfort noise packet should be constructed as The RTP header for the comfort noise packet should be constructed as
if the VAD noise were an independent codec, but sharing the media if the comfort noise were an independent codec. Thus, the RTP
clock and sequence number space with the associated voice codec. timestamp designates the beginning of the silence period. A static
Thus, the RTP timestamp designates the beginning of the silence payload type is assigned for a sampling rate of 8,000 Hz; if other
period, using the timestamp frequency of the payload type immediately sampling rates are needed, they should be defined through dynamic
preceding the CN packet. The RTP packet should not have the marker payload types. The RTP packet should not have the marker bit set.
bit set.
Note: dBrnc0 is the noise power measured in dBrnC, but referenced to
the zero-level transmission level point (TLP). Typically, the two-
wire interface in telephony is at the zero-level TLP of 0 dBm. dBrnC
is the power level of noise with C-message weighting expressed in
decibels relative to reference noise. Reference noise power is -90
Index Noise Level (dBrncO)
_____________________________
0 Idle Code
1 16.6
2 19.7
3 22.6
4 24.9
5 26.9
6 29.0
7 31.0
8 32.8
9 34.6
10 36.2
11 37.9
12 39.7
13 41.6
14 43.8
15 46.6
Table 2: G.764 noise level mapping
dBm or 1 pW. (dBm is the power level in decibels relative to 1 mW, The CN payload type is primarily for use with L16, DVI4, PCMA, PCMU
with an impedance of 600 Ohms.) The C-message weighting is described and other audio codecs that do not support comfort noise as part of
in [6]. To obtain dBmC0 levels, subtract 90 dB from the values the codec itself. G.723.1 and G.729 have their own comfort noise
listed. systems as part of Annexes A (G.723.1) and B (G.729), respectively.
4.5.3 DVI4 4.5.3 DVI4
DVI4 is specified, with pseudo-code, in [7] as the IMA ADPCM wave DVI4 is specified, with pseudo-code, in [6] as the IMA ADPCM wave
type. type.
However, the encoding defined here as DVI4 differs in three respects However, the encoding defined here as DVI4 differs in three respects
from this recommendation: from this recommendation:
o The header contains the predicted value rather than the first o The header contains the predicted value rather than the first
sample value. sample value.
o IMA ADPCM blocks contain an odd number of samples, since the o IMA ADPCM blocks contain an odd number of samples, since the
first sample of a block is contained just in the header first sample of a block is contained just in the header
skipping to change at page 12, line 18 skipping to change at page 12, line 13
sample encoding. sample encoding.
The "header" word for each channel has the following structure: The "header" word for each channel has the following structure:
int16 predict; /* predicted value of first sample int16 predict; /* predicted value of first sample
from the previous block (L16 format) */ from the previous block (L16 format) */
u_int8 index; /* current index into stepsize table */ u_int8 index; /* current index into stepsize table */
u_int8 reserved; /* set to zero by sender, ignored by receiver */ u_int8 reserved; /* set to zero by sender, ignored by receiver */
Each octet following the header contains two 4-bit samples, thus the Each octet following the header contains two 4-bit samples, thus the
number of samples per packet must be even.. number of samples per packet must be even.
Packing of samples for multiple channels is for further study. Packing of samples for multiple channels is for further study.
The document IMA Recommended Practices for Enhancing Digital Audio The document IMA Recommended Practices for Enhancing Digital Audio
Compatibility in Multimedia Systems (version 3.0) contains the Compatibility in Multimedia Systems (version 3.0) contains the
algorithm description. It is available from algorithm description. It is available from
Interactive Multimedia Association Interactive Multimedia Association
48 Maryland Avenue, Suite 202 48 Maryland Avenue, Suite 202
Annapolis, MD 21401-8011 Annapolis, MD 21401-8011
USA USA
phone: +1 410 626-1380 phone: +1 410 626-1380
4.5.4 G721 4.5.4 G722
G721 is specified in ITU recommendation G.721. Reference
implementations for G.721 are available as part of the CCITT/ITU-T
Software Tool Library (STL) from the ITU General Secretariat, Sales
Service, Place du Nations, CH-1211 Geneve 20, Switzerland. The
library is covered by a license.
4.5.5 G722
G722 is specified in ITU-T recommendation G.722, "7 kHz audio-coding G722 is specified in ITU-T Recommendation G.722, "7 kHz audio-coding
within 64 kbit/s". within 64 kbit/s".
4.5.6 G723 4.5.5 G723
G.723.1 is specified in ITU recommendation G.723.1, "Dual-rate speech G.723.1 is specified in ITU Recommendation G.723.1, "Dual-rate speech
coder for multimedia communications transmitting at 5.3 and 6.3 coder for multimedia communications transmitting at 5.3 and 6.3
kbit/s". Audio is encoded in 30 ms frames, with an additional delay kbit/s". The G.723.1 5.3/6.3 kbit/s codec was defined by the ITU-T as
of 7.5 ms due to look-ahead. A G.723.1 frame can be one of three a mandatory codec for ITU-T H.324 GSTN videophone terminal
sizes: 24 octets (6.3 kb/s frame), 20 octets (5.3 kb/s frame), or 4 applications. The algorithm has a floating point specification in
octets. These 4-octet frames are called SID frames (Silence Annex B to G.723.1, a silence compression algorithm in Annex A to
Insertion Descriptor) and are used to specify comfort noise G.723.1 and an encoded signal bit-error sensitivity specification in
G.723.1 Annex C.
This Recommendation specifies a coded representation that can be used
for compressing the speech signal component of multi-media services
at a very low bit rate. Audio is encoded in 30 ms frames, with an
additional delay of 7.5 ms due to look-ahead. A G.723.1 frame can be
one of three sizes: 24 octets (6.3 kb/s frame), 20 octets (5.3 kb/s
frame), or 4 octets. These 4-octet frames are called SID frames
(Silence Insertion Descriptor) and are used to specify comfort noise
parameters. There is no restriction on how 4, 20, and 24 octet frames 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 are intermixed. The least significant two bits of the first octet in
the frame determine the frame size and codec type: the frame determine the frame size and codec type:
bits content octets/frame bits content octets/frame
00 high-rate speech (6.3 kb/s) 24 00 high-rate speech (6.3 kb/s) 24
01 low-rate speech (5.3 kb/s) 20 01 low-rate speech (5.3 kb/s) 20
10 SID frame 4 10 SID frame 4
11 reserved 11 reserved
It is possible to switch between the two rates at any 30 ms frame 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 boundary. Both (5.3 kb/s and 6.3 kb/s) rates are a mandatory part of
the encoder and decoder. the encoder and decoder. This coder was optimized to represent speech
with near-toll quality at the above rates using a limited amount of
complexity.
4.5.7 G726-32 All the bits of the encoded bit stream are transmitted always from
the the least significant bit towards the most significant bit.
4.5.6 G726-16, G726-24, G726-32, G726-40
ITU-T Recommendation G.726 describes, among others, the algorithm 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 recommended for conversion of a single 64 kbit/s A-law or mu-law PCM
channel encoded at 8000 samples/sec to and from a 32 kbit/s channel. channel encoded at 8000 samples/sec to and from a 32 kbit/s channel.
The conversion is applied to the PCM stream using an Adaptive The conversion is applied to the PCM stream using an Adaptive
Differential Pulse Code Modulation (ADPCM) transcoding technique. Differential Pulse Code Modulation (ADPCM) transcoding technique.
G.726 is a backwards-compatible superset of G.721, a recommendation G.726 describes codecs operating at 16 kb/s (2 bits/sample), 24 kb/s
which is no longer in force. G.726 also describes codecs operating at (3 bits/sample), 32 kb/s (4 bits/sample), 40 kb/s (5 bits/sample).
40 (5 bits/sample), 24 (3 bits/sample) and 16 kb/s (2 bits/sample). These encodings are labeled G726-16, G726-24, G726-32 and G726-40,
These are labeled G726-40, G726-24 and G726-16, respectively. respectively.
Note: In 1990, ITU-T Recommendation G.721 was merged with
Recommendation G.723 into ITU-T Recommendation G.726. Thus, G726-32
designates the same algorithm as G721 in RFC 1890.
No header information shall be included as part of the audio data. No header information shall be included as part of the audio data.
The 4-bit code words of the G.726 encoding MUST be packed into octets The 4-bit code words of the G726-32 encoding MUST be packed into
as follows: the first code word is placed in the four least octets as follows: the first code word is placed in the four least
significant bits of the first octet, with the least significant bit significant bits of the first octet, with the least significant bit
of the code word in the least significant bit of the octet; the of the code word in the least significant bit of the octet; the
second code word is placed in the four most significant bits of the second code word is placed in the four most significant bits of the
first octet, with the most significant bit of the code word in the first octet, with the most significant bit of the code word in the
most significant bit of the octet. Subsequent pairs of the code words most significant bit of the octet. Subsequent pairs of the code words
shall be packed in the same way into successive octets, with the shall be packed in the same way into successive octets, with the
first code word of each pair placed in the least significant four first code word of each pair placed in the least significant four
bits of the octet. It is prefered that the voice sample be extended bits of the octet. It is prefered that the voice sample be extended
with silence such that the encoded value comprises an even number of with silence such that the encoded value comprises an even number of
code words. code words.
4.5.7 G727-16, G727-24, G727-32, G727-40
ITU-T Recommendation G.727, "5-, 4-, 3- and 2-bits sample embedded
adaptive differential pulse code modulation (ADPCM)", specifies an
embedded ADPCM algorithm which has the intrinsic capability of
dropping bits in the encoded words to alleviate network congestion
conditions. The algorithm, although not bitstream compatible with
G.726, was based and has a structure similar to the G.726 ADPCM
algorithm.
4.5.8 G728 4.5.8 G728
G728 is specified in ITU-T recommendation G.728, "Coding of speech at
G728 is specified in ITU-T Recommendation G.728, "Coding of speech at
16 kbit/s using low-delay code excited linear prediction". 16 kbit/s using low-delay code excited linear prediction".
A G.278 encoder translates 5 consecutive audio samples into a 10-bit 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 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 at 8,000 samples per second. The group of five consecutive samples is
called a vector. Four consecutive vectors, labeled V1-V4 (where V1 is called a vector. Four consecutive vectors, labeled V1 to V4 (where V1
to be played first by the receiver), build one G.728 frame. The four is to be played first by the receiver), build one G.728 frame. The
vectors of 40 bits are packed into 5 octets, labeled B1 through B5. 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 Referring to the figure below, the principle for bit order is
"maintenance of bit significance". Bits from an older vector are more "maintenance of bit significance". Bits from an older vector are more
significant than bits from newer vectors. The MSB of the frame goes 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. to the MSB of B1 and the LSB of the frame goes to LSB of B5. For
example: octet B1 contains the eight most significant bits of vector
V1, the MSB of V1 is MSB of B1.
1 2 3 3 1 2 3 3
0 0 0 0 9 0 0 0 0 9
++++++++++++++++++++++++++++++++++++++++ ++++++++++++++++++++++++++++++++++++++++
<---V1---><---V2---><---V3---><---V4---> <---V1---><---V2---><---V3---><---V4---> vectors
<--B1--><--B2--><--B3--><--B4--><--B5--> <--B1--><--B2--><--B3--><--B4--><--B5--> octets
<--------------Frame 1-----------------> <------------- frame 1 ---------------->
In particular, B1 contains the eight most significant bits of V1, 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 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, 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 and the six most significant bits of V2. B1 shall be placed first in
the RTP packet and B5 last. the RTP packet and B5 last.
4.5.9 G729 4.5.9 G729
G.729 and G.729A are defined in ITU-T Recommendation G.729, "Coding G729 is specified in ITU-T Recommendation G.729, "Coding of speech at
of Speech at 8 kbit/s using Conjugate Structure-Algebraic Code 8 kbit/s using conjugate structure-algebraic code excited linear
Excited Linear Predictive (CS-ACELP) Coding" and its Annex A, prediction (CS-ACELP)". A complexity-reduced version of the G.729
respectively. These two audio codecs are compatible with each other algorithm is specified in Annex A to Rec. G.729. The speech coding
on the wire so there is no need to distinguish further between them. algorithms in the main body of G.729 and in G.729 Annex A are fully
The codecs were optimized to represent speech with a high quality; interoperable with each other, so there is no need to further
G.729A achieves this with very low complexity. 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 [7].
A voice activity detector (VAD) and comfort noise generator (CNG) is A voice activity detector (VAD) and comfort noise generator (CNG)
defined in G.729 Annex B (G.729B). It can be used in conjunction with algorithm in Annex B of G.729 is recommended for digital simultaneous
either G.729 or G.729A. A G.729 or G.729A frame contains 10 octets, voice and data applications and can be used in conjunction with G.729
while the G.729B comfort noise frame contains 4 octets. 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:
An RTP packet may consist of zero or more G.729 or G.729A frames, 0 1
followed by zero or one G.729B payload. 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|L| LSF1 | LSF2 | GAIN |R|
|S| | | |E|
|F|0 1 2 3 4|0 1 2 3|0 1 2 3 4|S|
|0| | | |V| RESV = Reserved (zero)
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
An RTP packet may consist of zero or more G.729 or G.729 Annex A
frames, followed by zero or one G.729 Annex B payloads. The presence
of a comfort noise frame can be deduced from the length of the RTP
payload.
A floating-point version of the G.729, G.729 Annex A, and G.729 Annex
B will be available shortly as Annex C to Recommendation G.729.
The transmitted parameters of a G.729/G.729A 10-ms frame, consisting 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. of 80 bits, are defined in Recommendation G.729, Table 8/G.729.
The mapping of the these parameters is given below. Bits are numbered The mapping of the these parameters is given below. Bits are numbered
as Internet order, that is, the most significant bit is bit 0. as Internet order, that is, the most significant bit is bit 0.
0 1 2 3 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 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 | |L| L1 | L2 | L3 | P1 |P| C1 |
|0| | | | |0| | |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| | |0 1 2 3 4 5 6|0 1 2 3 4|0 1 2 3 4|0 1 2 3 4 5 6 7| |0 1 2 3 4|
| | | | | | | | | | | | | | | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
4 5 6
2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| C1 | S1 | GA1 | GB1 | P2 | C2 | | C1 | S1 | GA1 | GB1 | P2 | C2 |
| | | | | | | | | | | | | |
|5 6 7 8 9 1 1 1|3 2 1 0|2 1 0|3 2 1 0|4 3 2 1 0|0 1 2 3 4 5 6 7| |5 6 7 8 9 1 1 1|0 1 2 3|0 1 2|0 1 2 3|0 1 2 3 4|0 1 2 3 4 5 6 7|
| 0 1 2| | | | | | | 0 1 2| | | | | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
7
4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| C2 | S2 | GA2 | GB2 | | C2 | S2 | GA2 | GB2 |
| | | | | | | | | |
|8 9 1 1 1|0 1 2 3|0 1 2|0 1 2 3| |8 9 1 1 1|0 1 2 3|0 1 2|0 1 2 3|
| 0 1 2| | | | | 0 1 2| | | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
4.5.10 GSM 4.5.10 GSM
GSM (group speciale mobile) denotes the European GSM 06.10 GSM (group speciale mobile) denotes the European GSM 06.10
provisional standard for full-rate speech transcoding, prI-ETS 300 provisional standard for full-rate speech transcoding, prI-ETS 300
036, which is based on RPE/LTP (residual pulse excitation/long term 036, which is based on RPE/LTP (residual pulse excitation/long term
prediction) coding at a rate of 13 kb/s [8,9,10]. The standard can be prediction) coding at a rate of 13 kb/s [8,9,10]. The text of the
obtained from standard can be obtained from
ETSI (European Telecommunications Standards Institute) ETSI (European Telecommunications Standards Institute)
ETSI Secretariat: B.P.152 ETSI Secretariat: B.P.152
F-06561 Valbonne Cedex F-06561 Valbonne Cedex
France France
Phone: +33 92 94 42 00 Phone: +33 92 94 42 00
Fax: +33 93 65 47 16 Fax: +33 93 65 47 16
Blocks of 160 audio samples are compressed into 33 octets, for an Blocks of 160 audio samples are compressed into 33 octets, for an
effective data rate of 13,200 b/s. effective data rate of 13,200 b/s.
skipping to change at page 16, line 38 skipping to change at page 17, line 49
4.5.11 L8 4.5.11 L8
L8 denotes linear audio data, using 8-bits of precision with an L8 denotes linear audio data, using 8-bits of precision with an
offset of 128, that is, the most negative signal is encoded as zero. offset of 128, that is, the most negative signal is encoded as zero.
4.5.12 L16 4.5.12 L16
L16 denotes uncompressed audio data, using 16-bit signed L16 denotes uncompressed audio data, using 16-bit signed
representation with 65535 equally divided steps between minimum and representation with 65535 equally divided steps between minimum and
maximum signal level, ranging from maximum signal level, ranging from --32768 to 32767. The value is
to
represented in two's complement notation and network byte order.
4.5.13 LPC
LPC designates an experimental linear predictive encoding contributed
by Ron Frederick, Xerox PARC, which is based on an implementation
field field name bits field field name bits field field name bits field field name bits
__________________________________________________________ __________________________________________________________
1 LARc[0] 6 39 xmc[22] 3 1 LARc[0] 6 39 xmc[22] 3
2 LARc[1] 6 40 xmc[23] 3 2 LARc[1] 6 40 xmc[23] 3
3 LARc[2] 5 41 xmc[24] 3 3 LARc[2] 5 41 xmc[24] 3
4 LARc[3] 5 42 xmc[25] 3 4 LARc[3] 5 42 xmc[25] 3
5 LARc[4] 4 43 Nc[2] 7 5 LARc[4] 4 43 Nc[2] 7
6 LARc[5] 4 44 bc[2] 2 6 LARc[5] 4 44 bc[2] 2
7 LARc[6] 3 45 Mc[2] 2 7 LARc[6] 3 45 Mc[2] 2
8 LARc[7] 3 46 xmaxc[2] 6 8 LARc[7] 3 46 xmaxc[2] 6
skipping to change at page 17, line 45 skipping to change at page 18, line 45
30 xmc[13] 3 68 xmc[43] 3 30 xmc[13] 3 68 xmc[43] 3
31 xmc[14] 3 69 xmc[44] 3 31 xmc[14] 3 69 xmc[44] 3
32 xmc[15] 3 70 xmc[45] 3 32 xmc[15] 3 70 xmc[45] 3
33 xmc[16] 3 71 xmc[46] 3 33 xmc[16] 3 71 xmc[46] 3
34 xmc[17] 3 72 xmc[47] 3 34 xmc[17] 3 72 xmc[47] 3
35 xmc[18] 3 73 xmc[48] 3 35 xmc[18] 3 73 xmc[48] 3
36 xmc[19] 3 74 xmc[49] 3 36 xmc[19] 3 74 xmc[49] 3
37 xmc[20] 3 75 xmc[50] 3 37 xmc[20] 3 75 xmc[50] 3
38 xmc[21] 3 76 xmc[51] 3 38 xmc[21] 3 76 xmc[51] 3
Table 3: Ordering of GSM variables Table 2: Ordering of GSM variables
written by Ron Zuckerman, Motorola, posted to the Usenet group represented in two's complement notation and network byte order.
comp.dsp on June 26, 1992. The codec generates 14 octets for every
frame. The framesize is set to 20 ms, resulting in a bit rate of
5,600 b/s.
4.5.13 LPC
Octet Bit 0 Bit 1 Bit 2 Bit 3 Bit 4 Bit 5 Bit 6 Bit 7 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 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 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 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 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 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 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 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 7 xmaxc05 xmc0.0 xmc0.1 xmc0.2 xmc1.0 xmc1.1 xmc1.2 xmc2.0
skipping to change at page 18, line 41 skipping to change at page 19, line 40
24 xmc33.2 xmc34.0 xmc34.1 xmc34.2 xmc35.0 xmc35.1 xmc35.2 xmc36.0 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 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 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 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 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 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 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 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 32 xmc49.1 xmc49.2 xmc50.0 xmc50.1 xmc50.2 xmc51.0 xmc51.1 xmc51.2
LPC designates an experimental linear predictive encoding contributed
by Ron Frederick, Xerox PARC, which is based on an implementation
written by Ron Zuckerman, Motorola, posted to the Usenet group
comp.dsp on June 26, 1992. The codec generates 14 octets for every
frame. The framesize is set to 20 ms, resulting in a bit rate of
5,600 b/s.
4.5.14 MPA 4.5.14 MPA
MPA denotes MPEG-I or MPEG-II audio encapsulated as elementary MPA denotes MPEG-I or MPEG-II audio encapsulated as elementary
streams. The encoding is defined in ISO standards ISO/IEC 11172-3 streams. The encoding is defined in ISO standards ISO/IEC 11172-3
and 13818-3. The encapsulation is specified in RFC 2038 [11]. and 13818-3. The encapsulation is specified in RFC 2038 [11].
Sampling rate and channel count are contained in the payload. MPEG-I Sampling rate and channel count are contained in the payload. MPEG-I
audio supports sampling rates of 32000, 44100, and 48000 Hz (ISO/IEC audio supports sampling rates of 32000, 44100, and 48000 Hz (ISO/IEC
11172-3, section 1.1; "Scope"). MPEG-II additionally supports ISO/IEC 11172-3, section 1.1; "Scope"). MPEG-II additionally supports ISO/IEC
11172-3 Audio..."). 11172-3 Audio...").
4.5.15 PCMA 4.5.15 PCMA and PCMU
PCMA is specified in CCITT/ITU-T recommendation G.711. Audio data is
encoded as eight bits per sample, after logarithmic scaling. Code to
convert between linear and A-law companded data is available in [7].
A detailed description is given by Jayant and Noll [12].
4.5.16 PCMU
PCMU is specified in CCITT/ITU-T recommendation G.711. Audio data is PCMA and PCMU are specified in ITU-T Recommendation G.711. Audio data
encoded as eight bits per sample, after logarithmic scaling. Code to is encoded as eight bits per sample, after logarithmic scaling. PCMU
convert between linear and mu-law companded data is available in [7]. denotes mu-law scaling, PCMA A-law scaling. A detailed description is
PCMU is the encoding used for the Internet media type audio/basic. A given by Jayant and Noll [12]. Each G.711 octet shall be octet-
detailed description is given by Jayant and Noll [12]. aligned in an RTP packet. The sign bit of each G.711 octet shall
correspond to the most significant bit of the octet in the RTP packet
(i.e., assuming the G.711 samples are handled as octets on the host
machine, the sign bit shall be the most signficant bit of the octet
as defined by the host machine format). The 56 kb/s and 48 kb/s modes
of G.711 are not applicable to RTP, since G.711 shall always be
transmitted as 8-bit samples.
4.5.17 RED 4.5.16 RED
The redundant audio payload format "RED" is specified by RFC XXX. It The redundant audio payload format "RED" is specified by RFC XXX. It
defines a means by which multiple redundant copies of an audio packet 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 may be transmitted in a single RTP stream. Each packet in such a
stream contains, in addition to the audio data for that packetization stream contains, in addition to the audio data for that packetization
interval, a (more heavily compressed) copy of the data from the interval, a (more heavily compressed) copy of the data from the
previous packetization interval. This allows an approximation of the previous packetization interval. This allows an approximation of the
data from lost packets to be recovered upon decoding of the following data from lost packets to be recovered upon decoding of the following
packet, giving much improved sound quality when compared with silence packet, giving much improved sound quality when compared with silence
substitution for lost packets. substitution for lost packets.
4.5.18 VDVI 4.5.17 SX7300P
The SX7300P is a low-complexity CELP-based audio codec operating at a
sampling rate of 8000 Hz. It encodes blocks of 120 audio samples (15
ms) into an encoded frame of 14 octets, yielding an encoded bit rate
of approximately 7467 b/s.
4.5.18 SX8300P
The SX8300P is a low-complexity CELP-based audio codec operating at a
sampling rate of 8000 Hz. It encodes blocks of 120 audio samples (15
ms) into an encoded frame of 16 octets, yielding an encoded bit rate
of approximately 8533 b/s.
4.5.19 VDVI
VDVI is a variable-rate version of DVI4, yielding speech bit rates of 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 between 10 and 25 kb/s. It is specified for single-channel operation
only. Samples are packed into octets starting at the most- only. Samples are packed into octets starting at the most-significant
significant bit. bit.
It uses the following encoding: It uses the following encoding:
DVI4 codeword VDVI bit pattern DVI4 codeword VDVI bit pattern
_________________________________ _________________________________
0 00 0 00
1 010 1 010
2 1100 2 1100
3 11100 3 11100
4 111100 4 111100
skipping to change at page 20, line 28 skipping to change at page 21, line 47
The CELL-B encoding is a proprietary encoding proposed by Sun The CELL-B encoding is a proprietary encoding proposed by Sun
Microsystems. The byte stream format is described in RFC 2029 [13]. Microsystems. The byte stream format is described in RFC 2029 [13].
5.2 JPEG 5.2 JPEG
The encoding is specified in ISO Standards 10918-1 and 10918-2. The The encoding is specified in ISO Standards 10918-1 and 10918-2. The
RTP payload format is as specified in RFC 2035 [14]. RTP payload format is as specified in RFC 2035 [14].
5.3 H261 5.3 H261
The encoding is specified in CCITT/ITU-T standard H.261. The The encoding is specified in ITU-T Recommendation H.261, "Video codec
packetization and RTP-specific properties are described in RFC 2032 for audiovisual services at p x 64 kbit/s". The packetization and
[15]. RTP-specific properties are described in RFC 2032 [15].
5.4 MPV 5.4 H263
The encoding is specified in ITU-T Recommendation H.263, "Video
coding for low bit rate communication". The packetization and RTP-
specific properties are described in [16].
5.5 MPV
MPV designates the use MPEG-I and MPEG-II video encoding elementary MPV designates the use MPEG-I and MPEG-II video encoding elementary
streams as specified in ISO Standards ISO/IEC 11172 and 13818-2, streams as specified in ISO Standards ISO/IEC 11172 and 13818-2,
respectively. The RTP payload format is as specified in RFC 2038 respectively. The RTP payload format is as specified in RFC 2038
[11], Section 3. [11], Section 3.
5.5 MP2T 5.6 MP2T
MP2T designates the use of MPEG-II transport streams, for either MP2T designates the use of MPEG-II transport streams, for either
audio or video. The encapsulation is described in RFC 2038 [11], audio or video. The encapsulation is described in RFC 2038 [11],
Section 2. See the description of the MPA audio encoding for contact Section 2. See the description of the MPA audio encoding for contact
information. information.
5.6 nv 5.7 nv
The encoding is implemented in the program 'nv', version 4, developed The encoding is implemented in the program 'nv', version 4, developed
at Xerox PARC by Ron Frederick. Further information is available from at Xerox PARC by Ron Frederick. Further information is available from
the author: the author:
Ron Frederick Ron Frederick
Xerox Palo Alto Research Center Xerox Palo Alto Research Center
3333 Coyote Hill Road 3333 Coyote Hill Road
Palo Alto, CA 94304 Palo Alto, CA 94304
United States United States
electronic mail: frederic@parc.xerox.com electronic mail: frederic@parc.xerox.com
6 Payload Type Definitions 6 Payload Type Definitions
Table 4 defines this profile's static payload type values for the PT Table 3 defines this profile's static payload type values for the PT
field of the RTP data header. A new RTP payload format specification field of the RTP data header. A new RTP payload format specification
may be registered with the IANA by name, and may also be assigned a may be registered with the IANA by name, and may also be assigned a
static payload type value from the range marked in Section 3. static payload type value from the range marked in Section 3.
In addition, payload type values in the range In addition, payload type values in the range 96--127 may be defined
dynamically through a conference control protocol, which is beyond
may be defined dynamically through a conference control protocol, the scope of this document. For example, a session directory could
which is beyond the scope of this document. For example, a session specify that for a given session, payload type 96 indicates PCMU
directory could specify that for a given session, payload type 96 encoding, 8,000 Hz sampling rate, 2 channels. The payload type range
indicates PCMU encoding, 8,000 Hz sampling rate, 2 channels. The marked 'reserved' has been set aside so that RTCP and RTP packets can
payload type range marked 'reserved' has been set aside so that RTCP be reliably distinguished (see Section "Summary of Protocol
and RTP packets can be reliably distinguished (see Section "Summary Constants" of the RTP protocol specification).
of Protocol Constants" of the RTP protocol specification).
An RTP source emits a single RTP payload type at any given instant. An RTP source emits a single RTP payload type at any given instant.
The interleaving or multiplexing of several RTP media types within a The interleaving or multiplexing of several RTP media types within a
single RTP session is not allowed, but multiple RTP sessions may be single RTP session is not allowed, but multiple RTP sessions may be
used in parallel to send multiple media types. An RTP source may used in parallel to send multiple media types. An RTP source may
change payload types during a session. change payload types during a session.
The payload types currently defined in this profile are assigned to The payload types currently defined in this profile are assigned to
exactly one of three categories or media types : audio only, video exactly one of three categories or media types : audio only, video
only and those combining audio and video. A single RTP session only and those combining audio and video. A single RTP session
consists of payload types of one and only media type. consists of payload types of one and only media type.
Session participants agree through mechanisms beyond the scope of Session participants agree through mechanisms beyond the scope of
this specification on the set of payload types allowed in a given this specification on the set of payload types allowed in a given
session. This set may, for example, be defined by the capabilities session. This set may, for example, be defined by the capabilities
of the applications used, negotiated by a conference control protocol of the applications used, negotiated by a conference control protocol
or established by agreement between the human participants. The media or established by agreement between the human participants. The media
types in Table 4 are marked as "A" for audio, "V" for video and "AV" types in Table 3 are marked as "A" for audio, "V" for video and "AV"
for combined audio/video streams. for combined audio/video streams.
Audio applications operating under this profile should, at minimum, Audio applications operating under this profile should, at minimum,
be able to send and receive payload types 0 (PCMU) and 5 (DVI4). This be able to send and receive payload types 0 (PCMU) and 5 (DVI4). This
allows interoperability without format negotiation and successful allows interoperability without format negotiation and successful
negotation with a conference control protocol. negotation with a conference control protocol.
All current video encodings use a timestamp frequency of 90,000 Hz, All current video encodings use a timestamp frequency of 90,000 Hz,
the same as the MPEG presentation time stamp frequency. This the same as the MPEG presentation time stamp frequency. This
frequency yields exact integer timestamp increments for the typical frequency yields exact integer timestamp increments for the typical
skipping to change at page 22, line 19 skipping to change at page 23, line 43
and 50, 59.94 and 60 Hz field rates. While 90 kHz is the recommended 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 rate for future video encodings used within this profile, other rates
are possible. However, it is not sufficient to use the video frame are possible. However, it is not sufficient to use the video frame
rate (typically between 15 and 30 Hz) because that does not provide rate (typically between 15 and 30 Hz) because that does not provide
adequate resolution for typical synchronization requirements when adequate resolution for typical synchronization requirements when
calculating the RTP timestamp corresponding to the NTP timestamp in calculating the RTP timestamp corresponding to the NTP timestamp in
an RTCP SR packet. The timestamp resolution must also be sufficient an RTCP SR packet. The timestamp resolution must also be sufficient
for the jitter estimate contained in the receiver reports. for the jitter estimate contained in the receiver reports.
The standard video encodings and their payload types are listed in The standard video encodings and their payload types are listed in
Table 4. Table 3.
7 Port Assignment
As specified in the RTP protocol definition, RTP data is to be
carried on an even UDP port number and the corresponding RTCP packets
are to be carried on the next higher (odd) port number.
Applications operating under this profile may use any such UDP port
pair. For example, the port pair may be allocated randomly by a
session management program. A single fixed port number pair cannot be
required because multiple applications using this profile are likely
to run on the same host, and there are some operating systems that do
not allow multiple processes to use the same UDP port with different
multicast addresses.
However, port numbers 5004 and 5005 have been registered for use with
this profile for those applications that choose to use them as the
default pair. Applications that operate under multiple profiles may
use this port pair as an indication to select this profile if they
are not subject to the constraint of the previous paragraph.
Applications need not have a default and may require that the port
pair be explicitly specified. The particular port numbers were chosen
to lie in the range above 5000 to accomodate port number allocation
practice within the Unix operating system, where port numbers below
1024 can only be used by privileged processes and port numbers
between 1024 and 5000 are automatically assigned by the operating
system.
8 Bibliography 7 RTP over TCP and Similar Byte Stream Protocols
[1] Apple Computer, "Audio interchange file format AIFF-C," Aug. Under special circumstances, it may be necessary to carry RTP in
1991. (also ftp://ftp.sgi.com/sgi/aiff-c.9.26.91.ps.Z). protocols offering a byte stream abstraction, such as TCP, possibly
multiplexed with other data. If the application does not define its
own method of delineating RTP and RTCP packets, it SHOULD prefix each
packet with a two-octet length field.
PT encoding media type clock rate channels PT encoding media type clock rate channels
name (Hz) (audio) name (Hz) (audio)
_______________________________________________________________ _______________________________________________________________
0 PCMU A 8000 1 0 PCMU A 8000 1
1 1016 A 8000 1 1 1016 A 8000 1
2 G721 A 8000 1 2 G726-32 A 8000 1
3 GSM A 8000 1 3 GSM A 8000 1
4 G.723.1 A 8000 1 4 G723 A 8000 1
5 DVI4 A 8000 1 5 DVI4 A 8000 1
6 DVI4 A 16000 1 6 DVI4 A 16000 1
7 LPC A 8000 1 7 LPC A 8000 1
8 PCMA A 8000 1 8 PCMA A 8000 1
9 G722 A 16000 1 9 G722 A 16000 1
10 L16 A 44100 2 10 L16 A 44100 2
11 L16 A 44100 1 11 L16 A 44100 1
12 G723 A 8000 1 12 unassigned A
13 CN A 13 unassigned A
14 MPA A 90000 (see text) 14 MPA A 90000 (see text)
15 G728 A 8000 1 15 G728 A 8000 1
16 DVI4 A 11025 1 16 DVI4 A 11025 1
17 DVI4 A 22050 1 17 DVI4 A 22050 1
18 G729 A 8000 1 18 G729 A 8000 1
19--22 unassigned A 19 CN A 8000 1
20 unassigned A
21 unassigned A
22 unassigned A
23 unassigned A
24 unassigned V 24 unassigned V
25 CelB V 90000 25 CelB V 90000
26 JPEG V 90000 26 JPEG V 90000
27 unassigned V 27 unassigned V
28 nv V 90000 28 nv V 90000
29 unassigned V 29 unassigned V
30 unassigned V 30 unassigned V
31 H261 V 90000 31 H261 V 90000
32 MPV V 90000 32 MPV V 90000
33 MP2T AV 90000 33 MP2T AV 90000
34 H263 V 90000 34 H263 V 90000
35--71 unassigned ? 35--71 unassigned ?
72--76 reserved N/A N/A N/A 72--76 reserved N/A N/A N/A
77 RED A N/A N/A 77 RED A N/A N/A
78--95 unassigned ? 78--95 unassigned ?
96--127 dynamic ? 96--127 dynamic ?
Table 4: Payload types (PT) for standard audio and video encodings Table 3: Payload types (PT) for standard audio and video encodings
(Note: RTSP [17] provides its own encapsulation and does not need an
extra length indication.)
8 Port Assignment
As specified in the RTP protocol definition, RTP data is to be
carried on an even UDP or TCP port number and the corresponding RTCP
packets are to be carried on the next higher (odd) port number.
Applications operating under this profile may use any such UDP or TCP
port pair. For example, the port pair may be allocated randomly by a
session management program. A single fixed port number pair cannot be
required because multiple applications using this profile are likely
to run on the same host, and there are some operating systems that do
not allow multiple processes to use the same UDP port with different
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 accomodate port number allocation
practice within the Unix operating system, where port numbers below
1024 can only be used by privileged processes and port numbers
between 1024 and 5000 are automatically assigned by the operating
system.
9 Bibliography
[1] Apple Computer, "Audio interchange file format AIFF-C," Aug.
1991. (also ftp://ftp.sgi.com/sgi/aiff-c.9.26.91.ps.Z).
[2] Office of Technology and Standards, "Telecommunications: Analog [2] Office of Technology and Standards, "Telecommunications: Analog
to digital conversion of radio voice by 4,800 bit/second code excited to digital conversion of radio voice by 4,800 bit/second code excited
linear prediction (celp)," Federal Standard FS-1016, GSA, Room 6654; linear prediction (celp)," Federal Standard FS-1016, GSA, Room 6654;
7th & D Street SW; Washington, DC 20407 (+1-202-708-9205), 1990. 7th & D Street SW; Washington, DC 20407 (+1-202-708-9205), 1990.
[3] J. P. Campbell, Jr., T. E. Tremain, and V. C. Welch, "The [3] J. P. Campbell, Jr., T. E. Tremain, and V. C. Welch, "The
proposed Federal Standard 1016 4800 bps voice coder: CELP," Speech proposed Federal Standard 1016 4800 bps voice coder: CELP," Speech
Technology , vol. 5, pp. 58--64, April/May 1990. Technology , vol. 5, pp. 58--64, April/May 1990.
[4] J. P. Campbell, Jr., T. E. Tremain, and V. C. Welch, "The federal [4] J. P. Campbell, Jr., T. E. Tremain, and V. C. Welch, "The federal
standard 1016 4800 bps CELP voice coder," Digital Signal Processing , standard 1016 4800 bps CELP voice coder," Digital Signal Processing ,
vol. 1, no. 3, pp. 145--155, 1991. vol. 1, no. 3, pp. 145--155, 1991.
[5] J. P. Campbell, Jr., T. E. Tremain, and V. C. Welch, "The dod 4.8 [5] J. P. Campbell, Jr., T. E. Tremain, and V. C. Welch, "The dod 4.8
kbps standard (proposed federal standard 1016)," in Advances in kbps standard (proposed federal standard 1016)," in Advances in
Speech Coding (B. Atal, V. Cuperman, and A. Gersho, eds.), ch. 12, Speech Coding (B. Atal, V. Cuperman, and A. Gersho, eds.), ch. 12,
pp. 121--133, Kluwer Academic Publishers, 1991. pp. 121--133, Kluwer Academic Publishers, 1991.
[6] J. Bellamy, Digital Telephony New York: John Wiley & Sons, 1991. [6] IMA Digital Audio Focus and Technical Working Groups,
[7] IMA Digital Audio Focus and Technical Working Groups,
"Recommended practices for enhancing digital audio compatibility in "Recommended practices for enhancing digital audio compatibility in
multimedia systems (version 3.00)," tech. rep., Interactive multimedia systems (version 3.00)," tech. rep., Interactive
Multimedia Association, Annapolis, Maryland, Oct. 1992. Multimedia Association, Annapolis, Maryland, Oct. 1992.
[7] D. Deléam and J.-P. Petit, "Real-time implementations of the
recent ITU-T low bit rate speech coders on the TI TMS320C54X DSP:
results, methodology, and applications," in Proc. of International
Conference on Signal Processing, Technology, and Applications
(ICSPAT) , (Boston, Massachusetts), pp. 1656--1660, Oct. 1996.
[8] M. Mouly and M.-B. Pautet, The GSM system for mobile [8] M. Mouly and M.-B. Pautet, The GSM system for mobile
communications Lassay-les-Chateaux, France: Europe Media Duplication, communications Lassay-les-Chateaux, France: Europe Media Duplication,
1993. 1993.
[9] J. Degener, "Digital speech compression," Dr. Dobb's Journal , [9] J. Degener, "Digital speech compression," Dr. Dobb's Journal ,
Dec. 1994. Dec. 1994.
[10] S. M. Redl, M. K. Weber, and M. W. Oliphant, An Introduction to [10] S. M. Redl, M. K. Weber, and M. W. Oliphant, An Introduction to
GSM Boston: Artech House, 1995. GSM Boston: Artech House, 1995.
skipping to change at page 25, line 6 skipping to change at page 26, line 51
Internet Engineering Task Force, Oct. 1996. Internet Engineering Task Force, Oct. 1996.
[14] L. Berc, W. Fenner, R. Frederick, and S. McCanne, "RTP payload [14] L. Berc, W. Fenner, R. Frederick, and S. McCanne, "RTP payload
format for JPEG-compressed video," Request for Comments (Proposed format for JPEG-compressed video," Request for Comments (Proposed
Standard) RFC 2035, Internet Engineering Task Force, Oct. 1996. Standard) RFC 2035, Internet Engineering Task Force, Oct. 1996.
[15] T. Turletti and C. Huitema, "RTP payload format for H.261 video [15] T. Turletti and C. Huitema, "RTP payload format for H.261 video
streams," Request for Comments (Proposed Standard) RFC 2032, Internet streams," Request for Comments (Proposed Standard) RFC 2032, Internet
Engineering Task Force, Oct. 1996. Engineering Task Force, Oct. 1996.
9 Acknowledgements [16] C. C. Zhu, "RTP payload format for H.263 video streams,"
Internet Draft, Internet Engineering Task Force, Mar. 1997. Work in
progress.
The comments and careful review of Steve Casner are gratefully [17] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming
acknowledged. The GSM description was adopted from the IMTC Voice protocol (RTSP)," Internet Draft, Internet Engineering Task Force,
over IP Forum Service Interoperability Implementation Agreement July 1997. Work in progress.
(January 1997). Fred Burg helped with the G.729 description.
10 Address of Author 10 Acknowledgements
The comments and careful review of Steve Casner, Simao Campos and
Richard Cox are gratefully acknowledged. The GSM description was
adopted from the IMTC Voice over IP Forum Service Interoperability
Implementation Agreement (January 1997). Fred Burg and Terry Lyons
helped with the G.729 description.
11 Address of Author
Henning Schulzrinne Henning Schulzrinne
Dept. of Computer Science Dept. of Computer Science
Columbia University Columbia University
1214 Amsterdam Avenue 1214 Amsterdam Avenue
New York, NY 10027 New York, NY 10027
USA USA
electronic mail: schulzrinne@cs.columbia.edu electronic mail: schulzrinne@cs.columbia.edu
Current Locations of Related Resources 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 UTF-8
Information on the UCS Transformation Format 8 (UTF-8) is available Information on the UCS Transformation Format 8 (UTF-8) is available
at at
http://www.stonehand.com/unicode/standard/utf8.html http://www.stonehand.com/unicode/standard/utf8.html
1016 1016
An implementation is available at The U.S. DoD's Federal-Standard-1016 based 4800 bps code excited
linear prediction voice coder version 3.2 (CELP 3.2) Fortran and C
simulation source codes are available for worldwide distribution at
no charge (on DOS diskettes, but configured to compile on Sun SPARC
stations) from: Bob Fenichel, National Communications System,
Washington, D.C. 20305, phone +1-703-692-2124, fax +1-703-746-4960.
An implementation is also available at
ftp://ftp.super.org/pub/speech/celp_3.2a.tar.Z ftp://ftp.super.org/pub/speech/celp_3.2a.tar.Z
DVI4 DVI4
An implementation is available from Jack Jansen at An implementation is available from Jack Jansen at
ftp://ftp.cwi.nl/local/pub/audio/adpcm.shar ftp://ftp.cwi.nl/local/pub/audio/adpcm.shar
G721 G722
An implementation is available at An implementation of the G.722 algorithm is available as part of the
ftp://gaia.cs.umass.edu/pub/hgschulz/ccitt/ccitt_tools.tar.Z ITU-T STL, described above.
G723 G723
Reference implementations for G.723.1 are available as part of the The reference C code implementation defining the G.723.1 algorithm
CCITT/ITU-T Software Tool Library (STL) from the ITU General and its Annexes A, B, and C are available as an integral part of
Secretariat, Sales Service, Place du Nations, CH-1211 Geneve 20, Recommendation G.723.1 from the ITU Sales Service, address listed
Switzerland. The library is covered by a license. above. Both the algorithm and C code are covered by a specific
license. The ITU-T Secretariat should be contacted to obtain such
licensing information.
The specification also contains C source code. Source code files are G726-16 through G726-40
available at
http://www4.itu.ch/itudoc/itu-t/rec/g/g700-799/g723-1/723disk1_32415.html G726-16 through G726-40 are specified in the ITU-T Recommendation
G.726, "40, 32, 24, and 16 kb/s Adaptive Differential Pulse Code
Modulation (ADPCM)". An implementation of the G.726 algorithm is
available as part of the ITU-T STL, described above.
and test vectors at G727-16 through G727-40
http://www4.itu.ch/itudoc/itu-t/rec/g/g700-799/g723-1/723disk2_32416.html
G727-16 through G727-40 are specified in the ITU-T Recommendation
G.727, "5-, 4-, 3-, and 2-bit/sample embedded adaptive differential
pulse code modulation". An implementation of the G.727 algorithm will
be available in a future release of the ITU-T STL, described above.
G729 G729
Reference implementations for G.729, G.729A and G.729B are available The reference C code implementation defining the G.729 algorithm and
as part of the ITU-T Software Tool Library from the ITU General its Annexes A and B are available as an integral part of
Secretariat, Sales Service, Place de Nations, CH-1211 Geneve 20, Recommendation G.729 from the ITU Sales Service, listed above. Both
Switzerland. The library is covered by a license. the algorithm and the C code are covered by a specific license. The
contact information for obtaining the license is listed in the C
code.
GSM GSM
A reference implementation was written by Carsten Borman and Jutta A reference implementation was written by Carsten Borman and Jutta
Degener (TU Berlin, Germany). It is available at Degener (TU Berlin, Germany). It is available at
ftp://ftp.cs.tu-berlin.de/pub/local/kbs/tubmik/gsm/ 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 LPC
An implementation is available at An implementation is available at
ftp://parcftp.xerox.com/pub/net-research/lpc.tar.Z 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 [6].
Table of Contents
1 Introduction ........................................ 2
2 RTP and RTCP Packet Forms and Protocol Behavior ..... 3
3 Registering Payload Types ........................... 5
4 Audio ............................................... 6
4.1 Encoding-Independent Rules .......................... 6
4.2 Operating Recommendations ........................... 7
4.3 Guidelines for Sample-Based Audio Encodings ......... 8
4.4 Guidelines for Frame-Based Audio Encodings .......... 8
4.5 Audio Encodings ..................................... 9
4.5.1 1016 ................................................ 9
4.5.2 CN .................................................. 9
4.5.3 DVI4 ................................................ 11
4.5.4 G722 ................................................ 12
4.5.5 G723 ................................................ 12
4.5.6 G726-16, G726-24, G726-32, G726-40 .................. 13
4.5.7 G727-16, G727-24, G727-32, G727-40 .................. 14
4.5.8 G728 ................................................ 14
4.5.9 G729 ................................................ 15
4.5.10 GSM ................................................. 16
4.5.10.1 General Packaging Issues ............................ 17
4.5.10.2 GSM variable names and numbers ...................... 17
4.5.11 L8 .................................................. 17
4.5.12 L16 ................................................. 17
4.5.13 LPC ................................................. 18
4.5.14 MPA ................................................. 19
4.5.15 PCMA and PCMU ....................................... 20
4.5.16 RED ................................................. 20
4.5.17 SX7300P ............................................. 20
4.5.18 SX8300P ............................................. 20
4.5.19 VDVI ................................................ 20
5 Video ............................................... 21
5.1 CelB ................................................ 21
5.2 JPEG ................................................ 21
5.3 H261 ................................................ 21
5.4 H263 ................................................ 22
5.5 MPV ................................................. 22
5.6 MP2T ................................................ 22
5.7 nv .................................................. 22
6 Payload Type Definitions ............................ 22
7 RTP over TCP and Similar Byte Stream Protocols ...... 23
8 Port Assignment ..................................... 25
9 Bibliography ........................................ 25
10 Acknowledgements .................................... 27
11 Address of Author ................................... 27
 End of changes. 

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