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Internet Engineering Task Force AVT WG
Internet Draft Schulzrinne
ietf-avt-profile-new-00.txt Columbia U.
March 26, 1997
Expires: September 9, 1997
RTP Profile for Audio and Video Conferences with Minimal Control
STATUS OF THIS MEMO
This document is an Internet-Draft. Internet-Drafts are working
documents of the Internet Engineering Task Force (IETF), its areas,
and its working groups. Note that other groups may also distribute
working documents as Internet-Drafts.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as ``work in progress''.
To learn the current status of any Internet-Draft, please check the
``1id-abstracts.txt'' listing contained in the Internet-Drafts Shadow
Directories on ftp.is.co.za (Africa), nic.nordu.net (Europe),
munnari.oz.au (Pacific Rim), ds.internic.net (US East Coast), or
ftp.isi.edu (US West Coast).
Distribution of this document is unlimited.
ABSTRACT
This memo 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.
The document also describes how audio and video data may
be carried within RTP. It defines a set of standard
encodings and their names when used within RTP. However,
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Internet Draft Profile March 26, 1997
the encoding definitions are independent of the
particular transport mechanism used. The descriptions
provide pointers to reference implementations and the
detailed standards. This document is meant as an aid for
implementors of audio, video and other real-time
multimedia applications.
Changes
This draft revises RFC 1890. It is fully backwards-compatible with
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 Standard..
Besides wording clarifications and filling in RFC numbers for payload
type definitions, this draft adds payload types 4, 13, 16, 17, 18 and
34. The PostScript version of this draft contains change bars.
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
specification RFC submitted for DS status.
1 Introduction
This profile defines aspects of RTP left unspecified in the RTP
Version 2 protocol definition (RFC XXXX). This profile is intended
for the use within audio and video conferences with minimal session
control. In particular, no support for the negotiation of parameters
or membership control is provided. The profile is expected to be
useful in sessions where no negotiation or membership control are
used (e.g., using the static payload types and the membership
indications provided by RTCP), but this profile may also be useful in
conjunction with a higher-level control protocol.
Use of this profile occurs by use of the appropriate applications;
there is no explicit indication by port number, protocol identifier
or the like. Applications such as session directories should refer to
this profile as "RTP/AVP".
Other profiles may make different choices for the items specified
here.
This document also defines a set of payload formats for audio.
This draft defines the term media type as dividing encodings of audio
and video content into three classes: audio, video and audio/video
(interleaved).
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2 RTP and RTCP Packet Forms and Protocol Behavior
The section "RTP Profiles and Payload Format Specification" of RFC
TBD 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.
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 is sent every
reporting interval, other items should be sent only every third
reporting interval, with NAME sent seven out of eight times
within that slot and the remaining SDES items cyclically taking
up the eighth slot, as defined in Section 6.2.2 of the RTP
specification. In other words, NAME is sent in RTCP packets 1,
4, 7, 10, 13, 16, 19, while, say, EMAIL is used in RTCP packet
22.
Security: The RTP default security services are also the default
under this profile.
String-to-key mapping: A user-provided string ("pass phrase") is
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hashed with the MD5 algorithm to a 16-octet digest. An
-bit key is extracted from the digest by taking the first
bits from the digest. If several keys are needed with a total
length of 128 bits or less (as for triple DES), they are
extracted in order from that digest. The octet ordering is
specified in RFC 1423, Section 2.2. (Note that some DES
implementations require that the 56-bit key be expanded into 8
octets by inserting an odd parity bit in the most significant
bit of the octet to go with each 7 bits of the key.)
It is suggested that pass phrases are restricted to ASCII letters,
digits, the hyphen, and white space to reduce the the chance of
transcription errors when conveying keys by phone, fax, telex or
email.
The pass phrase may be preceded by a specification of the encryption
algorithm. Any characters up to the first slash (ASCII 0x2f) are
taken as the name of the encryption algorithm. The encryption format
specifiers should be drawn from RFC 1423 or any additional
identifiers registered with IANA. If no slash is present, DES-CBC is
assumed as default. The encryption algorithm specifier is case
sensitive.
The pass phrase typed by the user is transformed to a canonical form
before applying the hash algorithm. For that purpose, we define
return, tab, or vertical tab as well as all characters contained in
the Unicode space characters table. The transformation consists of
the following steps: (1) convert the input string to the ISO 10646
character set, using the UTF-8 encoding as specified in Annex P to
ISO/IEC 10646-1:1993 (ASCII characters require no mapping, but ISO
8859-1 characters do); (2) remove leading and trailing white space
characters; (3) replace one or more contiguous white space characters
by a single space (ASCII or UTF-8 0x20); (4) convert all letters to
lower case and replace sequences of characters and non-spacing
accents with a single character, where possible. A minimum length of
16 key characters (after applying the transformation) should be
enforced by the application, while applications must allow up to 256
characters of input.
Underlying protocol: The profile specifies the use of RTP over
unicast and multicast UDP. (This does not preclude the use of
these definitions when RTP is carried by other lower-layer
protocols.)
Transport mapping: The standard mapping of RTP and RTCP to
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transport-level addresses is used.
Encapsulation: No encapsulation of RTP packets is specified.
3 Registering Payload Types
This profile defines a set of standard encodings and their payload
types when used within RTP. Other encodings and their payload types
are to be registered with the Internet Assigned Numbers Authority
(IANA). When registering a new encoding/payload type, the following
information should be provided:
o name and description of encoding, in particular the RTP
timestamp clock rate; the names defined here are 3 or 4
characters long to allow a compact representation if needed;
o indication of who has change control over the encoding (for
example, ISO, CCITT/ITU, other international standardization
bodies, a consortium or a particular company or group of
companies);
o any operating parameters or profiles;
o a reference to a further description, if available, for
example (in order of preference) an RFC, a published paper, a
patent filing, a technical report, documented source code or a
computer manual;
o for proprietary encodings, contact information (postal and
email address);
o the payload type value for this profile, if necessary (see
below).
Note that not all encodings to be used by RTP need to be assigned a
static payload type. Non-RTP means beyond the scope of this memo
(such as directory services or invitation protocols) may be used to
establish a dynamic mapping between a payload type drawn from the
range
and an encoding. For implementor convenience, this profile contains
descriptions of encodings which do not currently have a static
payload type assigned to them.
Note that dynamic payload types should not be used without a well-
defined mechanism to indicate the mapping. Systems that expect to
interoperate with others operating under this profile should not
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assign proprietary encodings to particular, fixed payload types in
the range reserved for dynamic payload types.
The available payload type space is relatively small. Thus, new
static payload types are assigned only if the following conditions
are met:
o The encoding is of interest to the Internet community at
large.
o It offers benefits compared to existing encodings and/or is
required for interoperation with existing, widely deployed
conferencing or multimedia systems.
o The description is sufficient to build a decoder.
The four-character encoding names are those those by the Session
Description Protocol (SDP) (RFC XXXX) .
4 Audio
4.1 Encoding-Independent Rules
For applications which send no packets during silence, the first
packet of a talkspurt, that is, the first packet after a silence
period, is distinguished by setting the marker bit in the RTP data
header. The beginning of a talkspurt may be used to adjust the
playout delay to reflect changing network delays. Applications
without silence suppression set the bit to zero.
The RTP clock rate used for generating the RTP timestamp is
independent of the number of channels and the encoding; it equals the
number of sampling periods per second. For
-channel encodings, each sampling period (say,
of a second) generates
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.
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For more than two channels, the convention followed by the AIFF-C
audio interchange format should be followed [1], using the following
notation:
l left
r right
c center
S surround
F front
R rear
channels description channel
1 2 3 4 5 6
________________________________________________________________
2 stereo l r
3 l r c
4 quadrophonic Fl Fr Rl Rr
4 l c r S
5 Fl Fr Fc Sl Sr
6 l lc c r rc S
Samples for all channels belonging to a single sampling instant must
be within the same packet. The interleaving of samples from different
channels depends on the encoding. General guidelines are given in
Section 4.3 and 4.4.
The sampling frequency should be drawn from the set: 8000, 11025,
16000, 22050, 24000, 32000, 44100 and 48000 Hz. (The Apple Macintosh
computers have native sample rates of 22254.54 and 11127.27, which
can be converted to 22050 and 11025 with acceptable quality by
dropping 4 or 2 samples in a 20 ms frame.) However, most audio
encodings are defined for a more restricted set of sampling
frequencies. Receivers should be prepared to accept multi-channel
audio, but may choose to only play a single channel.
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.
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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
links with severe bandwidth constraints, a higher packetization delay
may be appropriate. A receiver should accept packets representing
between 0 and 200 ms of audio data. (For framed audio encodings, a
receiver should accept packets with 200 ms divided by the frame
duration, rounded up.) This restriction allows reasonable buffer
sizing for the receiver.
4.3 Guidelines for Sample-Based Audio Encodings
In sample-based encodings, each audio sample is represented by a
fixed number of bits. Within the compressed audio data, codes for
individual samples may span octet boundaries. An RTP audio packet may
contain any number of audio samples, subject to the constraint that
the number of bits per sample times the number of samples per packet
yields an integral octet count. Fractional encodings produce less
than one octet per sample.
The duration of an audio packet is determined by the number of
samples in the packet.
For sample-based encodings producing one or more octets per sample,
samples from different channels sampled at the same sampling instant
are packed in consecutive octets. For example, for a two-channel
encoding, the octet sequence is (left channel, first sample), (right
channel, first sample), (left channel, second sample), (right
channel, second sample), .... For multi-octet encodings, octets are
transmitted in network byte order (i.e., most significant octet
first).
The packing of sample-based encodings producing less than one octet
per sample is encoding-specific.
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 since the audio frame duration (in
octets) is defined as part of the encoding, as long as all frames
have the same length measured in octets. This does not work when
carrying frames of different sizes unless the frame sizes are
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relatively prime.
For frame-based codecs, the channel order is defined for the whole
block. That is, for two-channel audio, right and left samples are
coded independently, with the encoded frame for the left channel
preceding that for the right channel.
All frame-oriented audio codecs should be able to encode and decode
several consecutive frames within a single packet. Since the frame
size for the frame-oriented codecs is given, there is no need to use
a separate designation for the same encoding, but with different
number of frames per packet.
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 capturing time of the first sample in
the first frame, that is, the oldest information in the packet.
4.5 Audio Encodings
encoding sample/frame bits/sample ms/frame ms/packet
________________________________________________________________
1016 frame N/A 30 30
DVI4 sample 4 20
G721 sample 4 20
G722 sample 8 20
G723 frame N/A 30 30
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
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Encoding 1016 is a frame based encoding using code-excited linear
prediction (CELP) and is specified in Federal Standard FED-STD 1016
[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
The G.764-based VAD (voice activity detector) noise level packet
contains a single-octet message to the receiver to play comfort noise
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
period (which also indicates the transition from speech to silence),
but rate of noise level updates is implementation specific. The
mapping of the index to absolute noise levels measured on the
transmit side is given in Table 2, with the level index packed into
the least significant bits of the noise-level payload, as shown
below.
0
0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|0 0 0 0| level |
+-+-+-+-+-+-+-+-+
The RTP header for the comfort noise packet should be constructed as
if the VAD noise were an independent codec, but sharing the media
clock and sequence number space with the associated voice codec.
Thus, the RTP timestamp designates the beginning of the silence
period, using the timestamp frequency of the payload type immediately
preceding the CN packet. The RTP packet should not have the marker
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
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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,
with an impedance of 600 Ohms.) The C-message weighting is described
in [6]. To obtain dBmC0 levels, subtract 90 dB from the values
listed.
4.5.3 DVI4
DVI4 is specified, with pseudo-code, in [7] as the IMA ADPCM wave
type.
However, the encoding defined here as DVI4 differs in three respects
from this recommendation:
o The header contains the predicted value rather than the first
sample value.
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
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samples are packed in little-endian order.
Each packet contains a single DVI block. This profile only defines
the 4-bit-per-sample version, while IMA also specifies a 3-bit-per-
sample encoding.
The "header" word for each channel has the following structure:
int16 predict; /* predicted value of first sample
from the previous block (L16 format) */
u_int8 index; /* current index into stepsize table */
u_int8 reserved; /* set to zero by sender, ignored by receiver */
Each octet following the header contains two 4-bit samples, thus the
number of samples per packet must be even..
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
4.5.4 G721
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
within 64 kbit/s".
4.5.6 G723
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
kbit/s". Audio is encoded in 30 ms frames, with an additional delay
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of 7.5 ms due to look-ahead. A G.723.1 frame can be one of three
sizes: 24 octets (6.3 kb/s frame), 20 octets (5.3 kb/s frame), or 4
octets. These 4-octet frames are called SID frames (Silence
Insertion Descriptor) and are used to specify comfort noise
parameters. There is no restriction on how 4, 20, and 24 octet frames
are intermixed. The least significant two bits of the first octet in
the frame determine the frame size and codec type:
bits content octets/frame
00 high-rate speech (6.3 kb/s) 24
01 low-rate speech (5.3 kb/s) 20
10 SID frame 4
11 reserved
It is possible to switch between the two rates at any 30 ms frame
boundary. Both (5.3 kb/s and 6.3 kb/s) rates are a mandatory part of
the encoder and decoder.
4.5.7 G726-32
ITU-T Recommendation G.726 describes, among others, the algorithm
recommended for conversion of a single 64 kbit/s A-law or mu-law PCM
channel encoded at 8000 samples/sec to and from a 32 kbit/s channel.
The conversion is applied to the PCM stream using an Adaptive
Differential Pulse Code Modulation (ADPCM) transcoding technique.
G.726 is a backwards-compatible superset of G.721, a recommendation
which is no longer in force. G.726 also describes codecs operating at
40 (5 bits/sample), 24 (3 bits/sample) and 16 kb/s (2 bits/sample).
These are labeled G726-40, G726-24 and G726-16, respectively.
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
as follows: the first code word is placed in the four least
significant bits of the first octet, with the least significant bit
of the code word in the least significant bit of the octet; the
second code word is placed in the four most significant bits of the
first octet, with the most significant bit of the code word in the
most significant bit of the octet. Subsequent pairs of the code words
shall be packed in the same way into successive octets, with the
first code word of each pair placed in the least significant four
bits of the octet. It is prefered that the voice sample be extended
with silence such that the encoded value comprises an even number of
code words.
4.5.8 G728
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G728 is specified in ITU-T recommendation G.728, "Coding of speech at
16 kbit/s using low-delay code excited linear prediction".
A G.278 encoder translates 5 consecutive audio samples into a 10-bit
codebook index, resulting in a bit rate of 16 kb/s for audio sampled
at 8,000 samples per second. The group of five consecutive samples is
called a vector. Four consecutive vectors, labeled V1-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.
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--->
<--B1--><--B2--><--B3--><--B4--><--B5-->
<--------------Frame 1----------------->
In particular, B1 contains the eight most significant bits of V1,
with the MSB of V1 being the MSB of B1. B2 contains the two least
significant bits of V1, the more significant of the two in its MSB,
and the six most significant bits of V2. B1 shall be placed first in
the RTP packet and B5 last.
4.5.9 G729
G.729 and G.729A are defined in ITU-T Recommendation G.729, "Coding
of Speech at 8 kbit/s using Conjugate Structure-Algebraic Code
Excited Linear Predictive (CS-ACELP) Coding" and its Annex A,
respectively. These two audio codecs are compatible with each other
on the wire so there is no need to distinguish further between them.
The codecs were optimized to represent speech with a high quality;
G.729A achieves this with very low complexity.
A voice activity detector (VAD) and comfort noise generator (CNG) is
defined in G.729 Annex B (G.729B). It can be used in conjunction with
either G.729 or G.729A. A G.729 or G.729A frame contains 10 octets,
while the G.729B comfort noise frame contains 4 octets.
An RTP packet may consist of zero or more G.729 or G.729A frames,
followed by zero or one G.729B payload.
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The transmitted parameters of a G.729/G.729A 10-ms frame, consisting
of 80 bits, are defined in Recommendation G.729, Table 8/G.729.
The mapping of the these parameters is given below. Bits are numbered
as Internet order, that is, the most significant bit is bit 0.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|L| L1 | L2 | L3 | P1 |P| C1 |
|0| | | | |0| |
| |0 1 2 3 4 5 6|0 1 2 3 4|0 1 2 3 4|0 1 2 3 4 5 6 7| |0 1 2 3 4|
| | | | | | | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| 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|
| 0 1 2| | | | | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| C2 | S2 | GA2 | GB2 |
| | | | |
|8 9 1 1 1|0 1 2 3|0 1 2|0 1 2 3|
| 0 1 2| | | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
4.5.10 GSM
GSM (group speciale mobile) denotes the European GSM 06.10
provisional standard for full-rate speech transcoding, prI-ETS 300
036, which is based on RPE/LTP (residual pulse excitation/long term
prediction) coding at a rate of 13 kb/s [8,9,10]. 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
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effective data rate of 13,200 b/s.
4.5.10.1 General Packaging Issues
The GSM standard specifies the bit stream produced by the codec, but
does not specify how these bits should be packed for transmission.
Some software implementations of the GSM codec use a different
packing than that specified here.
In the GSM encoding used by RTP, the bits are 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 (4-7) and the 4 most significant bits of F1 (2-5) in the 4 least
significant bits (0-3). The second octet contains the 2 least bits of
F1 in bits 6-7, and F2 in bits 0-5, and so on. The order of the
fields in the frame is as follows:
4.5.10.2 GSM variable names and numbers
So if F.i signifies the ith bit of the field F, and bit 0 is the most
significant bit, and the bits of every octet are numbered from 0 to 7
from most to least significant, then in the RTP encoding we have:
4.5.11 L8
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.
4.5.12 L16
L16 denotes uncompressed audio data, using 16-bit signed
representation with 65535 equally divided steps between minimum and
maximum signal level, ranging from
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
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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 3: Ordering of GSM variables
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.
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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
4.5.14 MPA
MPA denotes MPEG-I or MPEG-II audio encapsulated as elementary
streams. The encoding is defined in ISO standards ISO/IEC 11172-3
and 13818-3. The encapsulation is specified in RFC 2038 [11].
Sampling rate and channel count are contained in the payload. MPEG-I
audio supports sampling rates of 32000, 44100, and 48000 Hz (ISO/IEC
11172-3, section 1.1; "Scope"). MPEG-II additionally supports ISO/IEC
11172-3 Audio...").
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4.5.15 PCMA
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
encoded as eight bits per sample, after logarithmic scaling. Code to
convert between linear and mu-law companded data is available in [7].
PCMU is the encoding used for the Internet media type audio/basic. A
detailed description is given by Jayant and Noll [12].
4.5.17 RED
The redundant audio payload format "RED" is specified by RFC XXX. 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 the
previous packetization interval. This allows an approximation of the
data from lost packets to be recovered upon decoding of the following
packet, giving much improved sound quality when compared with silence
substitution for lost packets.
4.5.18 VDVI
VDVI is a variable-rate version of DVI4, yielding speech bit rates of
between 10 and 25 kb/s. It is specified for single-channel operation
only. Samples are packed into octets starting at the most-
significant bit.
It uses the following encoding:
DVI4 codeword VDVI bit pattern
_________________________________
0 00
1 010
2 1100
3 11100
4 111100
5 1111100
6 11111100
7 11111110
8 10
9 011
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10 1101
11 11101
12 111101
13 1111101
14 11111101
15 11111111
5 Video
The following video encodings are currently defined, with their
abbreviated names used for identification:
5.1 CelB
The CELL-B encoding is a proprietary encoding proposed by Sun
Microsystems. The byte stream format is described in RFC 2029 [13].
5.2 JPEG
The encoding is specified in ISO Standards 10918-1 and 10918-2. The
RTP payload format is as specified in RFC 2035 [14].
5.3 H261
The encoding is specified in CCITT/ITU-T standard H.261. The
packetization and RTP-specific properties are described in RFC 2032
[15].
5.4 MPV
MPV designates the use MPEG-I and MPEG-II video encoding elementary
streams as specified in ISO Standards ISO/IEC 11172 and 13818-2,
respectively. The RTP payload format is as specified in RFC 2038
[11], Section 3.
5.5 MP2T
MP2T designates the use of MPEG-II transport streams, for either
audio or video. The encapsulation is described in RFC 2038 [11],
Section 2. See the description of the MPA audio encoding for contact
information.
5.6 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:
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Ron Frederick
Xerox Palo Alto Research Center
3333 Coyote Hill Road
Palo Alto, CA 94304
United States
electronic mail: frederic@parc.xerox.com
6 Payload Type Definitions
Table 4 defines this profile's static payload type values for the PT
field of the RTP data header. A new RTP payload format specification
may be registered with the IANA by name, and may also be assigned a
static payload type value from the range marked in Section 3.
In addition, payload type values in the range
may be defined dynamically through a conference control protocol,
which is beyond the scope of this document. For example, a session
directory could specify that for a given session, payload type 96
indicates PCMU encoding, 8,000 Hz sampling rate, 2 channels. The
payload type range marked 'reserved' has been set aside so that RTCP
and RTP packets can be reliably distinguished (see Section "Summary
of Protocol Constants" of the RTP protocol specification).
An RTP source emits a single RTP payload type at any given instant.
The interleaving or multiplexing of several RTP media types within a
single RTP session is not allowed, but multiple RTP sessions may be
used in parallel to send multiple media types. An RTP source may
change payload types during a session.
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. A single RTP session
consists of payload types of one and only media type.
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. The media
types in Table 4 are marked as "A" for audio, "V" for video and "AV"
for combined audio/video streams.
Audio applications operating under this profile should, at minimum,
be able to send and receive payload types 0 (PCMU) and 5 (DVI4). This
allows interoperability without format negotiation and successful
negotation with a conference control protocol.
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All current video encodings use a timestamp frequency of 90,000 Hz,
the same as the MPEG presentation time stamp frequency. This
frequency yields exact integer timestamp increments for the typical
24 (HDTV), 25 (PAL), and 29.97 (NTSC) and 30 Hz (HDTV) frame rates
and 50, 59.94 and 60 Hz field rates. While 90 kHz is the recommended
rate for future video encodings used within this profile, other rates
are possible. However, it is not sufficient to use the video frame
rate (typically between 15 and 30 Hz) because that does not provide
adequate resolution for typical synchronization requirements when
calculating the RTP timestamp corresponding to the NTP timestamp in
an RTCP SR packet. The timestamp resolution must also be sufficient
for the jitter estimate contained in the receiver reports.
The standard video encodings and their payload types are listed in
Table 4.
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
[1] Apple Computer, "Audio interchange file format AIFF-C," Aug.
1991. (also ftp://ftp.sgi.com/sgi/aiff-c.9.26.91.ps.Z).
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PT encoding media type clock rate channels
name (Hz) (audio)
_______________________________________________________________
0 PCMU A 8000 1
1 1016 A 8000 1
2 G721 A 8000 1
3 GSM A 8000 1
4 G.723.1 A 8000 1
5 DVI4 A 8000 1
6 DVI4 A 16000 1
7 LPC A 8000 1
8 PCMA A 8000 1
9 G722 A 16000 1
10 L16 A 44100 2
11 L16 A 44100 1
12 G723 A 8000 1
13 CN 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--22 unassigned A
24 unassigned V
25 CelB V 90000
26 JPEG V 90000
27 unassigned V
28 nv V 90000
29 unassigned V
30 unassigned V
31 H261 V 90000
32 MPV V 90000
33 MP2T AV 90000
34 H263 V 90000
35--71 unassigned ?
72--76 reserved N/A N/A N/A
77 RED A N/A N/A
78--95 unassigned ?
96--127 dynamic ?
Table 4: Payload types (PT) for standard audio and video encodings
[2] Office of Technology and Standards, "Telecommunications: Analog
to digital conversion of radio voice by 4,800 bit/second code excited
linear prediction (celp)," Federal Standard FS-1016, GSA, Room 6654;
7th & D Street SW; Washington, DC 20407 (+1-202-708-9205), 1990.
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[3] J. P. Campbell, Jr., T. E. Tremain, and V. C. Welch, "The
proposed Federal Standard 1016 4800 bps voice coder: CELP," Speech
Technology , vol. 5, pp. 58--64, April/May 1990.
[4] J. P. Campbell, Jr., T. E. Tremain, and V. C. Welch, "The federal
standard 1016 4800 bps CELP voice coder," Digital Signal Processing ,
vol. 1, no. 3, pp. 145--155, 1991.
[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
Speech Coding (B. Atal, V. Cuperman, and A. Gersho, eds.), ch. 12,
pp. 121--133, Kluwer Academic Publishers, 1991.
[6] J. Bellamy, Digital Telephony New York: John Wiley & Sons, 1991.
[7] 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.
[8] M. Mouly and M.-B. Pautet, The GSM system for mobile
communications Lassay-les-Chateaux, France: Europe Media Duplication,
1993.
[9] J. Degener, "Digital speech compression," Dr. Dobb's Journal ,
Dec. 1994.
[10] S. M. Redl, M. K. Weber, and M. W. Oliphant, An Introduction to
GSM Boston: Artech House, 1995.
[11] D. Hoffman, G. Fernando, and V. Goyal, "RTP payload format for
MPEG1/MPEG2 video," Request for Comments (Proposed Standard) RFC
2038, Internet Engineering Task Force, Oct. 1996.
[12] N. S. Jayant and P. Noll, Digital Coding of Waveforms--
Principles and Applications to Speech and Video Englewood Cliffs, New
Jersey: Prentice-Hall, 1984.
[13] 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.
[14] L. Berc, W. Fenner, R. Frederick, and S. McCanne, "RTP payload
format for JPEG-compressed video," Request for Comments (Proposed
Standard) RFC 2035, Internet Engineering Task Force, Oct. 1996.
[15] T. Turletti and C. Huitema, "RTP payload format for H.261 video
streams," Request for Comments (Proposed Standard) RFC 2032, Internet
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Engineering Task Force, Oct. 1996.
9 Acknowledgements
The comments and careful review of Steve Casner are gratefully
acknowledged. The GSM description was adopted from the IMTC Voice
over IP Forum Service Interoperability Implementation Agreement
(January 1997). Fred Burg helped with the G.729 description.
10 Address of Author
Henning Schulzrinne
Dept. of Computer Science
Columbia University
1214 Amsterdam Avenue
New York, NY 10027
USA
electronic mail: schulzrinne@cs.columbia.edu
Current Locations of Related Resources
UTF-8
Information on the UCS Transformation Format 8 (UTF-8) is available
at
http://www.stonehand.com/unicode/standard/utf8.html
1016
An implementation is available at
ftp://ftp.super.org/pub/speech/celp_3.2a.tar.Z
DVI4
An implementation is available from Jack Jansen at
ftp://ftp.cwi.nl/local/pub/audio/adpcm.shar
G721
An implementation is available at
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ftp://gaia.cs.umass.edu/pub/hgschulz/ccitt/ccitt_tools.tar.Z
G723
Reference implementations for G.723.1 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.
The specification also contains C source code. Source code files are
available at
http://www4.itu.ch/itudoc/itu-t/rec/g/g700-799/g723-1/723disk1_32415.html
and test vectors at
http://www4.itu.ch/itudoc/itu-t/rec/g/g700-799/g723-1/723disk2_32416.html
G729
Reference implementations for G.729, G.729A and G.729B are available
as part of the ITU-T Software Tool Library from the ITU General
Secretariat, Sales Service, Place de Nations, CH-1211 Geneve 20,
Switzerland. The library is covered by a license.
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/
LPC
An implementation is available at
ftp://parcftp.xerox.com/pub/net-research/lpc.tar.Z
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