draft-ietf-avt-profile-new-07.txt   draft-ietf-avt-profile-new-08.txt 
Internet Engineering Task Force AVT WG Internet Engineering Task Force AVT WG
Internet Draft Schulzrinne/Casner Internet Draft Schulzrinne/Casner
draft-ietf-avt-profile-new-07.txt Columbia U./Cisco Systems draft-ietf-avt-profile-new-08.txt Columbia U./Cisco Systems
October 21, 1999 January 14, 2000
Expires: April 21, 2000 Expires: July 14, 2000
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 and is in full conformance with This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026. all provisions of Section 10 of RFC2026.
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF), its areas, and its working groups. Note that Task Force (IETF), its areas, and its working groups. Note that
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bandwidth and separately for active senders and passive bandwidth and separately for active senders and passive
receivers. receivers.
o No specific action is taken in this document to address o No specific action is taken in this document to address
generic payload formats; it is assumed that if any generic generic payload formats; it is assumed that if any generic
payload formats are developed, they can be specified in payload formats are developed, they can be specified in
separate RFCs and that the session parameters they require for separate RFCs and that the session parameters they require for
operation can be specified in the MIME registration of those operation can be specified in the MIME registration of those
formats. formats.
o The specification of the CN (comfort noise) payload format
has been removed to a separate draft so that it may be
enhanced as a result of additional work in ITU-T. That draft
is intended for publication at Proposed Standard status.
Static payload type 13 is marked reserved here for the use of
that payload format (since CN has already been implemented
from earlier drafts of this profile). Static payload type 19
is also reserved because some revisions of the draft assigned
that number to CN to avoid an historic use of 13.
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) [1]. This profile is Version 2 protocol definition (RFC XXXX) [1]. This profile is
intended for the use within audio and video conferences with minimal intended for the use within audio and video conferences with minimal
session control. In particular, no support for the negotiation of session control. In particular, no support for the negotiation of
parameters or membership control is provided. The profile is expected parameters or membership control is provided. The profile is expected
to be useful in sessions where no negotiation or membership control to be useful in sessions where no negotiation or membership control
are used (e.g., using the static payload types and the membership are used (e.g., using the static payload types and the membership
indications provided by RTCP), but this profile may also be useful in indications provided by RTCP), but this profile may also be useful in
conjunction with a higher-level control protocol. conjunction with a higher-level control protocol.
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session control. In particular, no support for the negotiation of session control. In particular, no support for the negotiation of
parameters or membership control is provided. The profile is expected parameters or membership control is provided. The profile is expected
to be useful in sessions where no negotiation or membership control to be useful in sessions where no negotiation or membership control
are used (e.g., using the static payload types and the membership are used (e.g., using the static payload types and the membership
indications provided by RTCP), but this profile may also be useful in indications provided by RTCP), but this profile may also be useful in
conjunction with a higher-level control protocol. conjunction with a higher-level control protocol.
Use of this profile may be implicit in the use of the appropriate Use of this profile may be implicit in the use of the appropriate
applications; there may be no explicit indication by port number, applications; there may be no explicit indication by port number,
protocol identifier or the like. Applications such as session protocol identifier or the like. Applications such as session
directories should refer to this profile as "RTP/AVP". directories may use the name for this profile specified in Section 3.
Other profiles may make different choices for the items specified Other profiles may make different choices for the items specified
here. here.
This document also defines a set of encodings and payload formats for This document also defines a set of encodings and payload formats for
audio and video. audio and video.
1.1 Terminology 1.1 Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
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Underlying protocol: The profile specifies the use of RTP over Underlying protocol: The profile specifies the use of RTP over
unicast and multicast UDP as well as TCP. (This does not unicast and multicast UDP as well as TCP. (This does not
preclude the use of these definitions when RTP is carried preclude the use of these definitions when RTP is carried
by other lower-layer protocols.) by 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: A minimal TCP encapsulation is defined. Encapsulation: A minimal TCP encapsulation is defined.
3 Registering Additional Encodings with IANA 3 IANA Considerations
The RTP specification establishes a registry of profile names for use
by higher-level control protocols, such as the Session Description
Protocol (SDP), RFC 2327 [5], to refer to transport methods. This
profile registers the name "RTP/AVP".
3.1 Registering Additional Encodings
This profile lists a set of encodings, each of which is comprised of This profile lists a set of encodings, each of which is comprised of
a particular media data compression or representation plus a payload a particular media data compression or representation plus a payload
format for encapsulation within RTP. Some of those payload formats format for encapsulation within RTP. Some of those payload formats
are specified here, while others are specified in separate RFCs. It are specified here, while others are specified in separate RFCs. It
is expected that additional encodings beyond the set listed here will is expected that additional encodings beyond the set listed here will
be created in the future and specified in additional payload format be created in the future and specified in additional payload format
RFCs. RFCs.
This profile also assigns to each encoding a short name which MAY be This profile also assigns to each encoding a short name which MAY be
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registered name of the encoding/payload format, along with any registered name of the encoding/payload format, along with any
additional required parameters such as the RTP timestamp clock rate additional required parameters such as the RTP timestamp clock rate
and number of channels, to a payload type number. This association and number of channels, to a payload type number. This association
is effective only for the duration of the RTP session in which the is effective only for the duration of the RTP session in which the
dynamic payload type binding is made. This association applies only dynamic payload type binding is made. This association applies only
to the RTP session for which it is made, thus the numbers can be re- to the RTP session for which it is made, thus the numbers can be re-
used for different encodings in different sessions so the number used for different encodings in different sessions so the number
space limitation is avoided. space limitation is avoided.
This profile reserves payload type numbers in the range 96-127 This profile reserves payload type numbers in the range 96-127
exclusively for dynamic assignment. Applications should first use exclusively for dynamic assignment. Applications SHOULD first use
values in this range for dynamic payload types. Only applications values in this range for dynamic payload types. Those applications
which need to define more than 32 dynamic payload types MAY bind which need to define more than 32 dynamic payload types MAY bind
codes below 96, in which case it is RECOMMENDED that unassigned codes below 96, in which case it is RECOMMENDED that unassigned
payload type numbers be used first. However, the statically assigned payload type numbers be used first. However, the statically assigned
payload types are default bindings and MAY be dynamically bound to payload types are default bindings and MAY be dynamically bound to
new encodings if needed. Redefining payload types below 96 may cause new encodings if needed. Redefining payload types below 96 may cause
incorrect operation if an attempt is made to join a session without incorrect operation if an attempt is made to join a session without
obtaining session description information that defines the dynamic obtaining session description information that defines the dynamic
payload types. payload types.
Dynamic payload types SHOULD NOT be used without a well-defined Dynamic payload types SHOULD NOT be used without a well-defined
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sampling period (say, 1/8000 of a second) generates N samples. (This sampling period (say, 1/8000 of a second) generates N samples. (This
terminology is standard, but somewhat confusing, as the total number terminology is standard, but somewhat confusing, as the total number
of samples generated per second is then the sampling rate times the of samples generated per second is then the sampling rate times the
channel count.) 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 [6], using the following audio interchange format SHOULD be followed [6], using the following
notation: notation, unless some other convention is specified for a particular
encoding or payload format:
l left l left
r right r right
c center c center
S surround S surround
F front F front
R rear R rear
channels description channel channels description channel
1 2 3 4 5 6 1 2 3 4 5 6
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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 instant at which the first sample in The RTP timestamp reflects the instant at which the first sample in
the first frame was sampled, that is, the oldest information in the the first frame was sampled, that is, the oldest information in the
packet. packet.
4.5 Audio Encodings 4.5 Audio Encodings
The characteristics of the audio encodings described in this document
are shown in Table 1; they are listed in order of their payload type
in Table 4. While most audio codecs are only specified for a fixed
sampling rate, some sample-based algorithms (indicated by an entry of
"var." in the sampling rate column of Table 1) may be used with
different sampling rates, resulting in different coded bit rates.
When used with a sampling rate other than that for which a static
payload type is defined, non-RTP means beyond the scope of this memo
name of sampling default name of sampling default
encoding sample/frame bits/sample rate ms/frame ms/packet encoding sample/frame bits/sample rate ms/frame ms/packet
__________________________________________________________________ __________________________________________________________________
1016 frame N/A 8,000 30 30 1016 frame N/A 8,000 30 30
CN frame N/A var.
DVI4 sample 4 var. 20 DVI4 sample 4 var. 20
G722 sample 8 16,000 20 G722 sample 8 16,000 20
G723 frame N/A 8,000 30 30 G723 frame N/A 8,000 30 30
G726-32 sample 4 8,000 20 G726-32 sample 4 8,000 20
G728 frame N/A 8,000 2.5 20 G728 frame N/A 8,000 2.5 20
G729 frame N/A 8,000 10 20 G729 frame N/A 8,000 10 20
GSM frame N/A 8,000 20 20 GSM frame N/A 8,000 20 20
GSM-HR frame N/A 8,000 20 20 GSM-HR frame N/A 8,000 20 20
GSM-EFR frame N/A 8,000 20 20 GSM-EFR frame N/A 8,000 20 20
L8 sample 8 var. 20 L8 sample 8 var. 20
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LPC frame N/A 8,000 20 20 LPC frame N/A 8,000 20 20
MPA frame N/A var. var. MPA frame N/A var. var.
PCMA sample 8 var. 20 PCMA sample 8 var. 20
PCMU sample 8 var. 20 PCMU sample 8 var. 20
QCELP frame N/A 8,000 20 20 QCELP frame N/A 8,000 20 20
VDVI sample var. var. 20 VDVI sample var. var. 20
Table 1: Properties of Audio Encodings (N/A: not applicable; var.: Table 1: Properties of Audio Encodings (N/A: not applicable; var.:
variable) variable)
The characteristics of the audio encodings described in this document
are shown in Table 1; they are listed in order of their payload type
in Table 4. While most audio codecs are only specified for a fixed
sampling rate, some sample-based algorithms (indicated by an entry of
"var." in the sampling rate column of Table 1) may be used with
different sampling rates, resulting in different coded bit rates.
When used with a sampling rate other than that for which a static
payload type is defined, non-RTP means beyond the scope of this memo
MUST be used to define a dynamic payload type and MUST indicate the MUST be used to define a dynamic payload type and MUST indicate the
selected RTP timestamp clock rate, which is usually the same as the selected RTP timestamp clock rate, which is usually the same as the
sampling rate for audio. sampling rate for audio.
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
[7,8,9,10]. [7,8,9,10].
4.5.2 CN 4.5.2 DVI4
The CN (comfort noise) packet contains a single-octet message to the
receiver to play comfort noise at the absolute level specified. This
message would normally be sent once at the beginning of a silence
period (which also indicates the transition from speech to silence),
but the rate of noise level updates is implementation specific. The
magnitude of the noise level is packed into the least significant
bits of the noise-level payload, as shown below.
The noise level is expressed in -dBov, with values from 0 to 127
representing 0 to -127 dBov. dBov is the level relative to the
overload of the system. (Note: Representation relative to the
overload point of a system is particularly useful for digital
implementations, since one does not need to know the relative
calibration of the analog circuitry.) For example, in a 16-bit linear
PCM system (L16), a signal with 0 dBov represents a square wave with
the maximum possible amplitude (+/-32767), and -63 dBov corresponds
to -58 dBm0 in a standard telephone system. (dBm is the power level
in decibels relative to 1 mW, with an impedance of 600 Ohms.)
0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|0| level |
+-+-+-+-+-+-+-+-+
The RTP header for the comfort noise packet SHOULD be constructed as
if the comfort noise were an independent codec. Thus, the RTP
timestamp designates the beginning of the silence period. A static
payload type is assigned for a sampling rate of 8,000 Hz; if other
sampling rates are needed, they MUST be defined through dynamic
payload types. The RTP packet SHOULD NOT have the marker bit set.
The CN payload type is primarily for use with L16, DVI4, PCMA, PCMU
and other audio codecs that do not support comfort noise as part of
the codec itself. G.723.1 and G.729 have their own comfort noise
systems as part of Annexes A (G.723.1) and B (G.729), respectively.
4.5.3 DVI4
DVI4 is specified, with pseudo-code, in [11] as the IMA ADPCM wave DVI4 is specified, with pseudo-code, in [11] 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 RTP DVI4 header contains the predicted value rather than o The RTP DVI4 header contains the predicted value rather than
the first sample value contained the IMA ADPCM block header. the first sample value contained the IMA ADPCM block header.
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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 G722 4.5.3 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". The G.722 encoder produces a stream of octets, within 64 kbit/s". The G.722 encoder produces a stream of octets,
each of which SHALL be octet-aligned in an RTP packet. The first bit each of which SHALL be octet-aligned in an RTP packet. The first bit
transmitted in the G.722 octet, which is the most significant bit of transmitted in the G.722 octet, which is the most significant bit of
the higher sub-band sample, SHALL correspond to the most significant the higher sub-band sample, SHALL correspond to the most significant
bit of the octet in the RTP packet. bit of the octet in the RTP packet.
Even though the actual sampling rate for G.722 audio is 16000 Hz, the Even though the actual sampling rate for G.722 audio is 16000 Hz, the
RTP clock rate for the G722 payload format is 8000 Hz because that RTP clock rate for the G722 payload format is 8000 Hz because that
value was erroneously assigned in RFC 1890 and must remain unchanged value was erroneously assigned in RFC 1890 and must remain unchanged
for backward compatibility. The octet rate or sample-pair rate is for backward compatibility. The octet rate or sample-pair rate is
8000 Hz. 8000 Hz.
4.5.5 G723 4.5.4 G723
G723 is specified in ITU Recommendation G.723.1, "Dual-rate speech G723 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". The G.723.1 5.3/6.3 kbit/s codec was defined by the ITU-T as kbit/s". The G.723.1 5.3/6.3 kbit/s codec was defined by the ITU-T as
a mandatory codec for ITU-T H.324 GSTN videophone terminal a mandatory codec for ITU-T H.324 GSTN videophone terminal
applications. The algorithm has a floating point specification in applications. The algorithm has a floating point specification in
Annex B to G.723.1, a silence compression algorithm in Annex A to Annex B to G.723.1, a silence compression algorithm in Annex A to
G.723.1 and an encoded signal bit-error sensitivity specification in G.723.1 and an encoded signal bit-error sensitivity specification in
G.723.1 Annex C. G.723.1 Annex C.
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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. This coder was optimized to represent speech the encoder and decoder. This coder was optimized to represent speech
with near-toll quality at the above rates using a limited amount of with near-toll quality at the above rates using a limited amount of
complexity. complexity.
The packing of the encoded bit stream into octets and the The packing of the encoded bit stream into octets and the
transmission order of the octets is specified in G.723.1. transmission order of the octets is specified in G.723.1.
4.5.6 G726-32 4.5.5 G726-32
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 describes codecs operating at 16 kb/s (2 bits/sample), 24 kb/s G.726 describes codecs operating at 16 kb/s (2 bits/sample), 24 kb/s
(3 bits/sample), 32 kb/s (4 bits/sample), 40 kb/s (5 bits/sample). (3 bits/sample), 32 kb/s (4 bits/sample), 40 kb/s (5 bits/sample).
Packetization is specified here only for the 32 kb/s encoding which Packetization is specified here only for the 32 kb/s encoding which
is labeled G726-32. is labeled G726-32.
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significant bit of the code word in the least significant bit of the 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 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 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 word in the most significant bit of the octet. Subsequent pairs of
the code words SHALL be packed in the same way into successive 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 octets, with the first code word of each pair placed in the least
significant four bits of the octet. The number of samples per packet significant four bits of the octet. The number of samples per packet
MUST be even because there is no means to indicate a partially filled MUST be even because there is no means to indicate a partially filled
last octet. last octet.
4.5.7 G728 4.5.6 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 to V4 (where V1 called a vector. Four consecutive vectors, labeled V1 to V4 (where V1
is to be played first by the receiver), build one G.728 frame. The 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 four vectors of 40 bits are packed into 5 octets, labeled B1 through
B5. B1 SHALL be placed first in the RTP packet. B5. B1 SHALL be placed first in the RTP packet.
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<---V1---><---V2---><---V3---><---V4---> vectors <---V1---><---V2---><---V3---><---V4---> vectors
<--B1--><--B2--><--B3--><--B4--><--B5--> octets <--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.
4.5.8 G729 4.5.7 G729
G729 is specified in ITU-T Recommendation G.729, "Coding of speech at G729 is specified in ITU-T Recommendation G.729, "Coding of speech at
8 kbit/s using conjugate structure-algebraic code excited linear 8 kbit/s using conjugate structure-algebraic code excited linear
prediction (CS-ACELP)". A reduced-complexity version of the G.729 prediction (CS-ACELP)". A reduced-complexity version of the G.729
algorithm is specified in Annex A to Rec. G.729. The speech coding algorithm is specified in Annex A to Rec. G.729. The speech coding
algorithms in the main body of G.729 and in G.729 Annex A are fully algorithms in the main body of G.729 and in G.729 Annex A are fully
interoperable with each other, so there is no need to further interoperable with each other, so there is no need to further
distinguish between them. The G.729 and G.729 Annex A codecs were 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 optimized to represent speech with high quality, where G.729 Annex A
trades some speech quality for an approximate 50% complexity trades some speech quality for an approximate 50% complexity
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7 7
4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 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| | | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
4.5.9 GSM 4.5.8 GSM
GSM (group speciale mobile) denotes the European GSM 06.10 standard GSM (group speciale mobile) denotes the European GSM 06.10 standard
for full-rate speech transcoding, ETS 300 961, which is based on for full-rate speech transcoding, ETS 300 961, which is based on
RPE/LTP (residual pulse excitation/long term prediction) coding at a RPE/LTP (residual pulse excitation/long term prediction) coding at a
rate of 13 kb/s [13,14,15]. The text of the standard can be obtained rate of 13 kb/s [13,14,15]. The text of the standard can be obtained
from 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
skipping to change at page 18, line 4 skipping to change at page 17, line 24
RPE/LTP (residual pulse excitation/long term prediction) coding at a RPE/LTP (residual pulse excitation/long term prediction) coding at a
rate of 13 kb/s [13,14,15]. The text of the standard can be obtained rate of 13 kb/s [13,14,15]. The text of the standard can be obtained
from 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.
4.5.9.1 General Packaging Issues 4.5.8.1 General Packaging Issues
The GSM standard (ETS 300 961) specifies the bit stream produced by The GSM standard (ETS 300 961) specifies the bit stream produced by
the codec, but does not specify how these bits should be packed for the codec, but does not specify how these bits should be packed for
transmission. The packetization specified here has subsequently been transmission. The packetization specified here has subsequently been
adopted in ETSI Technical Specification TS 101 318. Some software adopted in ETSI Technical Specification TS 101 318. Some software
implementations of the GSM codec use a different packing than that implementations of the GSM codec use a different packing than that
specified here. specified here.
In the GSM packing used by RTP, the bits SHALL be packed beginning In the GSM packing used by RTP, the bits SHALL be packed beginning
from the most significant bit. Every 160 sample GSM frame is coded 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 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 bit signature (0xD), followed by the MSB encoding of the fields of
the frame. The first octet thus contains 1101 in the 4 most the frame. The first octet thus contains 1101 in the 4 most
significant bits (0-3) and the 4 most significant bits of F1 (0-3) in significant bits (0-3) and the 4 most significant bits of F1 (0-3) in
the 4 least significant bits (4-7). The second octet contains the 2 the 4 least significant bits (4-7). The second octet contains the 2
least significant bits of F1 in bits 0-1, and F2 in bits 2-7, and so least significant bits of F1 in bits 0-1, and F2 in bits 2-7, and so
on. The order of the fields in the frame is described in Table 2. on. The order of the fields in the frame is described in Table 2.
4.5.9.2 GSM variable names and numbers 4.5.8.2 GSM variable names and numbers
In the RTP encoding we have the bit pattern described in Table 3,
where F.i signifies the ith bit of the field F, bit 0 is the most
significant bit, and the bits of every octet are numbered from 0 to 7
from most to least significant.
4.5.10 GSM-HR
GSM-HR denotes GSM 06.20 half rate speech transcoding, specified in
ETS 300 969 which is available from ETSI at the address given in
Section 4.5.9. This codec has a frame length of 112 bits (14 octets).
Packing of the fields in the codec bit stream into octets for
transmission in RTP is done in a manner similar to that specified
here for the original GSM 06.10 codec and is specified in ETSI
Technical Specification TS 101 318.
4.5.11 GSM-EFR
GSM-EFR denotes GSM 06.60 enhanced full rate speech transcoding,
specified in ETS 300 969 which is available from ETSI at the address
given in Section 4.5.9. This codec has a frame length of 244 bits.
For transmission in RTP, each codec frame is packed into a 31 octet
(248 bit) buffer beginning with a 4-bit signature 0xC in a manner
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 19, line 47 skipping to change at page 18, line 47
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 2: Ordering of GSM variables Table 2: Ordering of GSM variables
similar to that specified here for the original GSM 06.10 codec. The In the RTP encoding we have the bit pattern described in Table 3,
packing is specified in ETSI Technical Specification TS 101 318. where F.i signifies the ith bit of the field F, bit 0 is the most
significant bit, and the bits of every octet are numbered from 0 to 7
from most to least significant.
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
skipping to change at page 20, line 43 skipping to change at page 19, line 44
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
Table 3: GSM payload format Table 3: GSM payload format
4.5.12 L8 4.5.9 GSM-HR
GSM-HR denotes GSM 06.20 half rate speech transcoding, specified in
ETS 300 969 which is available from ETSI at the address given in
Section 4.5.8. This codec has a frame length of 112 bits (14 octets).
Packing of the fields in the codec bit stream into octets for
transmission in RTP is done in a manner similar to that specified
here for the original GSM 06.10 codec and is specified in ETSI
Technical Specification TS 101 318.
4.5.10 GSM-EFR
GSM-EFR denotes GSM 06.60 enhanced full rate speech transcoding,
specified in ETS 300 969 which is available from ETSI at the address
given in Section 4.5.8. This codec has a frame length of 244 bits.
For transmission in RTP, each codec frame is packed into a 31 octet
(248 bit) buffer beginning with a 4-bit signature 0xC in a manner
similar to that specified here for the original GSM 06.10 codec. The
packing is specified in ETSI Technical Specification TS 101 318.
4.5.11 L8
L8 denotes linear audio data samples, using 8-bits of precision with L8 denotes linear audio data samples, using 8-bits of precision with
an offset of 128, that is, the most negative signal is encoded as an offset of 128, that is, the most negative signal is encoded as
zero. zero.
4.5.13 L16 4.5.12 L16
L16 denotes uncompressed audio data samples, using 16-bit signed L16 denotes uncompressed audio data samples, 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 -32768 to 32767. The value is maximum signal level, ranging from -32768 to 32767. The value is
represented in two's complement notation and transmitted in network represented in two's complement notation and transmitted in network
byte order (most significant byte first). byte order (most significant byte first).
4.5.14 LPC 4.5.13 LPC
LPC designates an experimental linear predictive encoding contributed LPC designates an experimental linear predictive encoding contributed
by Ron Frederick, Xerox PARC, which is based on an implementation by Ron Frederick, Xerox PARC, which is based on an implementation
written by Ron Zuckerman, Motorola, posted to the Usenet group written by Ron Zuckerman, Motorola, posted to the Usenet group
comp.dsp on June 26, 1992. The codec generates 14 octets for every 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 frame. The framesize is set to 20 ms, resulting in a bit rate of
5,600 b/s. 5,600 b/s.
4.5.15 MPA 4.5.14 MPA
MPA denotes MPEG-1 or MPEG-2 audio encapsulated as elementary MPA denotes MPEG-1 or MPEG-2 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 2250 [16]. and 13818-3. The encapsulation is specified in RFC 2250 [16].
The encoding may be at any of three levels of complexity, called The encoding may be at any of three levels of complexity, called
Layer I, II and III. The selected layer as well as the sampling rate Layer I, II and III. The selected layer as well as the sampling rate
and channel count are indicated in the payload. The RTP timestamp and channel count are indicated in the payload. The RTP timestamp
clock rate is always 90000, independent of the sampling rate. MPEG-1 clock rate is always 90000, independent of the sampling rate. MPEG-1
audio supports sampling rates of 32, 44.1, and 48 kHz (ISO/IEC audio supports sampling rates of 32, 44.1, and 48 kHz (ISO/IEC
11172-3, section 1.1; "Scope"). MPEG-2 supports sampling rates of 16, 11172-3, section 1.1; "Scope"). MPEG-2 supports sampling rates of 16,
22.05 and 24 kHz. The number of samples per frame is fixed, but the 22.05 and 24 kHz. The number of samples per frame is fixed, but the
frame size will vary with the sampling rate and bit rate. frame size will vary with the sampling rate and bit rate.
4.5.16 PCMA and PCMU 4.5.15 PCMA and PCMU
PCMA and PCMU are specified in ITU-T Recommendation G.711. Audio data PCMA and PCMU are specified in ITU-T Recommendation G.711. Audio data
is encoded as eight bits per sample, after logarithmic scaling. PCMU is encoded as eight bits per sample, after logarithmic scaling. PCMU
denotes mu-law scaling, PCMA A-law scaling. A detailed description is denotes mu-law scaling, PCMA A-law scaling. A detailed description is
given by Jayant and Noll [17]. Each G.711 octet SHALL be octet- given by Jayant and Noll [17]. Each G.711 octet SHALL be octet-
aligned in an RTP packet. The sign bit of each G.711 octet SHALL 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 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 (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 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 as defined by the host machine format). The 56 kb/s and 48 kb/s modes
of G.711 are not applicable to RTP, since PCMA and PCMU SHALL always of G.711 are not applicable to RTP, since PCMA and PCMU MUST always
be transmitted as 8-bit samples. be transmitted as 8-bit samples.
4.5.17 QCELP 4.5.16 QCELP
The Electronic Industries Association (EIA) & Telecommunications The Electronic Industries Association (EIA) & Telecommunications
Industry Association (TIA) standard IS-733, "TR45: High Rate Speech Industry Association (TIA) standard IS-733, "TR45: High Rate Speech
Service Option for Wideband Spread Spectrum Communications Systems," Service Option for Wideband Spread Spectrum Communications Systems,"
defines the QCELP audio compression algorithm for use in wireless defines the QCELP audio compression algorithm for use in wireless
CDMA applications. The QCELP CODEC compresses each 20 milliseconds of CDMA applications. The QCELP CODEC compresses each 20 milliseconds of
8000 Hz, 16- bit sampled input speech into one of four different size 8000 Hz, 16- bit sampled input speech into one of four different size
output frames: Rate 1 (266 bits), Rate 1/2 (124 bits), Rate 1/4 (54 output frames: Rate 1 (266 bits), Rate 1/2 (124 bits), Rate 1/4 (54
bits) or Rate 1/8 (20 bits). For typical speech patterns, this bits) or Rate 1/8 (20 bits). For typical speech patterns, this
results in an average output of 6.8 k bits/sec for normal mode and results in an average output of 6.8 k bits/sec for normal mode and
4.7 k bits/sec for reduced rate mode. The packetization of the QCELP 4.7 k bits/sec for reduced rate mode. The packetization of the QCELP
audio codec is described in [18]. audio codec is described in [18].
4.5.18 RED 4.5.17 RED
The redundant audio payload format "RED" is specified by RFC 2198 The redundant audio payload format "RED" is specified by RFC 2198
[19]. It defines a means by which multiple redundant copies of an [19]. It defines a means by which multiple redundant copies of an
audio packet may be transmitted in a single RTP stream. Each packet 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 in such a stream contains, in addition to the audio data for that
packetization interval, a (more heavily compressed) copy of the data packetization interval, a (more heavily compressed) copy of the data
from a previous packetization interval. This allows an approximation from a previous packetization interval. This allows an approximation
of the data from lost packets to be recovered upon decoding of a of the data from lost packets to be recovered upon decoding of a
subsequent packet, giving much improved sound quality when compared subsequent packet, giving much improved sound quality when compared
with silence substitution for lost packets. with silence substitution for lost packets.
4.5.19 VDVI 4.5.18 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 bit. The last octet is padded with 1 bits if the last significant bit. The last octet is padded with 1 bits if the last
sample does not fill the last octet. This padding is distinct from sample does not fill the last octet. This padding is distinct from
the valid codewords. The receiver needs to detect the padding the valid codewords. The receiver needs to detect the padding
because there is no explicit count of samples in the packet. because there is no explicit count of samples in the packet.
It uses the following encoding: It uses the following encoding:
skipping to change at page 24, line 23 skipping to change at page 23, line 47
The encoding is specified in ITU-T Recommendation H.261, "Video codec The encoding is specified in ITU-T Recommendation H.261, "Video codec
for audiovisual services at p x 64 kbit/s". The packetization and for audiovisual services at p x 64 kbit/s". The packetization and
RTP-specific properties are described in RFC 2032 [23]. RTP-specific properties are described in RFC 2032 [23].
5.5 H263 5.5 H263
The encoding is specified in the 1996 version of ITU-T Recommendation The encoding is specified in the 1996 version of ITU-T Recommendation
H.263, "Video coding for low bit rate communication". The H.263, "Video coding for low bit rate communication". The
packetization and RTP-specific properties are described in RFC 2190 packetization and RTP-specific properties are described in RFC 2190
[24]. [24]. The H263-1998 payload format is RECOMMENDED over this one for
use by new implementations.
5.6 H263-1998 5.6 H263-1998
The encoding is specified in the 1998 version of ITU-T Recommendation The encoding is specified in the 1998 version of ITU-T Recommendation
H.263, "Video coding for low bit rate communication". The H.263, "Video coding for low bit rate communication". The
packetization and RTP-specific properties are described in RFC 2429 packetization and RTP-specific properties are described in RFC 2429
[25]. Because the 1998 version of H.263 is a superset of the 1996 [25]. Because the 1998 version of H.263 is a superset of the 1996
syntax, this payload format can also be used with the 1996 version of syntax, this payload format can also be used with the 1996 version of
H.263, and is RECOMMENDED for this use by new implementations. This H.263, and is RECOMMENDED for this use by new implementations. This
payload format does not replace RFC 2190, which continues to be used payload format does not replace RFC 2190, which continues to be used
by existing implementations, and may be required for backward by existing implementations, and may be required for backward
compatibility in new implementations. Implementations using the new compatibility in new implementations. Implementations using the new
features of the 1998 version of H.263 MUST use the payload format features of the 1998 version of H.263 MUST use the payload format
skipping to change at page 25, line 45 skipping to change at page 25, line 22
6 Payload Type Definitions 6 Payload Type Definitions
Tables 4 and 5 define this profile's static payload type values for Tables 4 and 5 define this profile's static payload type values for
the PT field of the RTP data header. In addition, payload type the PT field of the RTP data header. In addition, payload type
values in the range 96-127 MAY be defined dynamically through a values in the range 96-127 MAY be defined dynamically through a
conference control protocol, which is beyond the scope of this conference control protocol, which is beyond the scope of this
document. For example, a session directory could specify that for a document. For example, a session directory could specify that for a
given session, payload type 96 indicates PCMU encoding, 8,000 Hz given session, payload type 96 indicates PCMU encoding, 8,000 Hz
sampling rate, 2 channels. Entries in Tables 4 and 5 with payload sampling rate, 2 channels. Entries in Tables 4 and 5 with payload
type "dyn" have no static payload type assigned and are only used type "dyn" have no static payload type assigned and are only used
with a dynamic payload type. The payload type range marked `reserved' with a dynamic payload type. Payload type 13 is reserved for a
has been set aside so that RTCP and RTP packets can be reliably comfort noise payload format to be specified in a separate RFC.
distinguished (see Section "Summary of Protocol Constants" of the RTP Payload type 19 is also marked "reserved" because some draft versions
protocol specification). of this specification assigned that number to a comfort noise payload
format. The payload type range 72-76 is marked "reserved" so that
RTCP and RTP packets can be reliably distinguished (see Section
"Summary of Protocol Constants" of the RTP protocol specification).
The payload types currently defined in this profile are assigned to 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. The media types are marked only and those combining audio and video. The media types are marked
in Tables 4 and 5 as "A", "V" and "AV", respectively. Payload types in Tables 4 and 5 as "A", "V" and "AV", respectively. Payload types
of different media types SHALL NOT be interleaved or multiplexed of different media types SHALL NOT be interleaved or multiplexed
within a single RTP session, but multiple RTP sessions MAY be used in within a single RTP session, but multiple RTP sessions MAY be used in
parallel to send multiple media types. An RTP source MAY change parallel to send multiple media types. An RTP source MAY change
payload types within the same media type during a session. See the payload types within the same media type during a session. See the
section "Multiplexing RTP Sessions" of RFC XXXX for additional section "Multiplexing RTP Sessions" of RFC XXXX for additional
skipping to change at page 26, line 23 skipping to change at page 26, line 5
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. or established by agreement between the human participants.
Audio applications operating under this profile SHOULD, at a minimum, Audio applications operating under this profile SHOULD, at a minimum,
be able to send and/or receive payload types 0 (PCMU) and 5 (DVI4). be able to send and/or receive payload types 0 (PCMU) and 5 (DVI4).
This allows interoperability without format negotiation and ensures This allows interoperability without format negotiation and ensures
successful negotation with a conference control protocol. successful negotation with a conference control protocol.
7 RTP over TCP and Similar Byte Stream Protocols
Under special circumstances, it may be necessary to carry RTP in
protocols offering a byte stream abstraction, such as TCP, possibly
multiplexed with other data. If the application does not define its
own method of delineating RTP and RTCP packets, it SHOULD prefix each
packet with a two-octet length field.
(Note: RTSP [27] provides its own encapsulation and does not need an
extra length indication.)
8 Port Assignment
As specified in the RTP protocol definition, RTP data SHOULD be
carried on an even UDP or TCP port number and the corresponding RTCP
packets SHOULD be carried on the next higher (odd) port number.
Applications operating under this profile MAY use any such UDP or TCP
port pair. For example, the port pair MAY be allocated randomly by a
session management program. A single fixed port number pair cannot be
required because multiple applications using this profile are likely
to run on the same host, and there are some operating systems that do
not allow multiple processes to use the same UDP port with different
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
PT encoding media type clock rate channels PT encoding media type clock rate channels
name (Hz) name (Hz)
___________________________________________________ ___________________________________________________
0 PCMU A 8000 1 0 PCMU A 8000 1
1 1016 A 8000 1 1 1016 A 8000 1
2 G726-32 A 8000 1 2 G726-32 A 8000 1
3 GSM A 8000 1 3 GSM A 8000 1
4 G723 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 8000 1 9 G722 A 8000 1
10 L16 A 44100 2 10 L16 A 44100 2
11 L16 A 44100 1 11 L16 A 44100 1
12 QCELP A 8000 1 12 QCELP A 8000 1
13 CN A 13 reserved 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 unassigned A 8000 1 19 reserved A
20 unassigned A 20 unassigned A
21 unassigned A 21 unassigned A
22 unassigned A 22 unassigned A
23 unassigned A 23 unassigned A
dyn GSM-HR A 8000 1 dyn GSM-HR A 8000 1
dyn GSM-EFR A 8000 1 dyn GSM-EFR A 8000 1
dyn RED A dyn L8 A var. var.
dyn RED A (see text)
dyn VDVI A var. 1
Table 4: Payload types (PT) for audio encodings Table 4: Payload types (PT) for audio encodings
default pair. Applications that operate under multiple profiles MAY 7 RTP over TCP and Similar Byte Stream Protocols
use this port pair as an indication to select this profile if they
are not subject to the constraint of the previous paragraph.
Applications need not have a default and MAY require that the port
pair be explicitly specified. The particular port numbers were chosen
to lie in the range above 5000 to accommodate port number allocation
practice within some versions of the Unix operating system, where
port numbers below 1024 can only be used by privileged processes and
port numbers between 1024 and 5000 are automatically assigned by the
operating system.
9 Changes from RFC 1890 Under special circumstances, it may be necessary to carry RTP in
protocols offering a byte stream abstraction, such as TCP, possibly
multiplexed with other data. If the application does not define its
own method of delineating RTP and RTCP packets, it SHOULD prefix each
packet with a two-octet length field in network order (most
significant octet first).
(Note: RTSP [27] provides its own encapsulation and does not need an
PT encoding media type clock rate PT encoding media type clock rate
name (Hz) name (Hz)
____________________________________________ ____________________________________________
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
skipping to change at page 28, line 30 skipping to change at page 27, line 30
77-95 unassigned ? 77-95 unassigned ?
96-127 dynamic ? 96-127 dynamic ?
dyn BT656 V 90000 dyn BT656 V 90000
dyn H263-1998 V 90000 dyn H263-1998 V 90000
dyn MP1S V 90000 dyn MP1S V 90000
dyn MP2P V 90000 dyn MP2P V 90000
dyn BMPEG V 90000 dyn BMPEG V 90000
Table 5: Payload types (PT) for video and combined encodings Table 5: Payload types (PT) for video and combined encodings
extra length indication.)
8 Port Assignment
As specified in the RTP protocol definition, RTP data SHOULD be
carried on an even UDP or TCP port number and the corresponding RTCP
packets SHOULD be carried on the next higher (odd) port number.
Applications operating under this profile MAY use any such UDP or TCP
port pair. For example, the port pair MAY be allocated randomly by a
session management program. A single fixed port number pair cannot be
required because multiple applications using this profile are likely
to run on the same host, and there are some operating systems that do
not allow multiple processes to use the same UDP port with different
multicast addresses.
However, port numbers 5004 and 5005 have been registered for use with
this profile for those applications that choose to use them as the
default pair. Applications that operate under multiple profiles MAY
use this port pair as an indication to select this profile if they
are not subject to the constraint of the previous paragraph.
Applications need not have a default and MAY require that the port
pair be explicitly specified. The particular port numbers were chosen
to lie in the range above 5000 to accommodate port number allocation
practice within some versions of the Unix operating system, where
port numbers below 1024 can only be used by privileged processes and
port numbers between 1024 and 5000 are automatically assigned by the
operating system.
9 Changes from RFC 1890
This RFC revises RFC 1890. It is fully backwards-compatible with RFC This RFC revises RFC 1890. It is fully backwards-compatible with RFC
1890 and codifies existing practice. The changes are listed below. 1890 and codifies existing practice. The changes are listed below.
o Additional payload formats and/or expanded descriptions were o Additional payload formats and/or expanded descriptions were
included for CN, G722, G723, G726, G728, G729, GSM, GSM-HR, included for G722, G723, G726, G728, G729, GSM, GSM-HR, GSM-
GSM-EFR, QCELP, RED, VDVI, BT656, H263-1998, MP1S, MP2P and EFR, QCELP, RED, VDVI, BT656, H263, H263-1998, MP1S, MP2P and
BMPEG. BMPEG.
o Static payload types 4, 12, 13, 16, 17, 18 and 34 were added. o Static payload types 4, 12, 16, 17, 18 and 34 were added, and
13 and 19 were reserved.
o The policy is established that no additional registration of o A new Section "IANA Considerations" was added to specify the
static payload types for this Profile will be made beyond regstration of the name for this profile and to establish a
those included in Tables 4 and 5, but additional encoding new policy that no additional registration of static payload
names may be registered as MIME subtypes. types for this profile will be made beyond those included in
Tables 4 and 5, but that additional encoding names may be
registered as MIME subtypes for binding to dynamic payload
types.
o In Section 4.1, the requirement level for setting of the o In Section 4.1, the requirement level for setting of the
marker bit on the first packet after silence for audio was marker bit on the first packet after silence for audio was
changed from "is" to "SHOULD be". changed from "is" to "SHOULD be".
o Similarly, text was added to specify that the marker bit o Similarly, text was added to specify that the marker bit
SHOULD be set to one on the last packet of a video frame, and SHOULD be set to one on the last packet of a video frame, and
that video frames are distinguished by their timestamps. that video frames are distinguished by their timestamps.
o This profile follows the suggestion in the RTP spec that RTCP o This profile follows the suggestion in the RTP spec that RTCP
skipping to change at page 29, line 48 skipping to change at page 29, line 36
allow the explanation of multiplexing RTP sessions in allow the explanation of multiplexing RTP sessions in
Section 6 to be more clear regarding the multiplexing of Section 6 to be more clear regarding the multiplexing of
multiple media. multiple media.
- The explanation of how to determine the number of audio - The explanation of how to determine the number of audio
frames in a packet from the length was expanded. frames in a packet from the length was expanded.
- More description of the allocation of bandwidth to SDES - More description of the allocation of bandwidth to SDES
items is given. items is given.
- A note was added that the convention for the order of
channels specified in Section 4.1 may be overridden by a
particular encoding or payload format specification.
- The terms MUST, SHOULD, MAY, etc. are used as defined in RFC - The terms MUST, SHOULD, MAY, etc. are used as defined in RFC
2119. 2119.
o A second author for this document was added. o A second author for this document was added.
10 Security Considerations 10 Security Considerations
Implementations using the profile defined in this specification are Implementations using the profile defined in this specification are
subject to the security considerations discussed in the RTP subject to the security considerations discussed in the RTP
specification [1]. This profile does not specify any different specification [1]. This profile does not specify any different
skipping to change at page 33, line 14 skipping to change at page 33, line 9
GSM Boston: Artech House, 1995. GSM Boston: Artech House, 1995.
[16] D. Hoffman, G. Fernando, V. Goyal, and M. Civanlar, "RTP payload [16] D. Hoffman, G. Fernando, V. Goyal, and M. Civanlar, "RTP payload
format for MPEG1/MPEG2 video," Request for Comments (Proposed format for MPEG1/MPEG2 video," Request for Comments (Proposed
Standard) RFC 2250, Internet Engineering Task Force, Jan. 1998. Standard) RFC 2250, Internet Engineering Task Force, Jan. 1998.
[17] N. S. Jayant and P. Noll, Digital Coding of Waveforms-- [17] N. S. Jayant and P. Noll, Digital Coding of Waveforms--
Principles and Applications to Speech and Video Englewood Cliffs, New Principles and Applications to Speech and Video Englewood Cliffs, New
Jersey: Prentice-Hall, 1984. Jersey: Prentice-Hall, 1984.
[18] K. McKay, "RTP Payload Format for PureVoice(tm) Audio", Internet [18] K. McKay, "RTP Payload Format for PureVoice(tm) Audio", Request
Draft, Internet Engineering Task Force, Oct. 1998. Work in progress. for Comments (Proposed Standard) RFC 2658, Internet Engineering Task
Force, Aug. 1999.
[19] C. Perkins, I. Kouvelas, O. Hodson, V. Hardman, M. Handley, J.C. [19] C. Perkins, I. Kouvelas, O. Hodson, V. Hardman, M. Handley, J.C.
Bolot, A. Vega-Garcia, and S. Fosse-Parisis, "RTP Payload for Bolot, A. Vega-Garcia, and S. Fosse-Parisis, "RTP Payload for
Redundant Audio Data," Request for Comments (Proposed Standard) RFC Redundant Audio Data," Request for Comments (Proposed Standard) RFC
2198, Internet Engineering Task Force, Sep. 1997. 2198, Internet Engineering Task Force, Sep. 1997.
[20] D. Tynan, "RTP payload format for BT.656 Video Encoding," [20] D. Tynan, "RTP payload format for BT.656 Video Encoding,"
Request for Comments (Proposed Standard) RFC 2431, Internet Request for Comments (Proposed Standard) RFC 2431, Internet
Engineering Task Force, Oct. 1998. Engineering Task Force, Oct. 1998.
skipping to change at page 36, line 10 skipping to change at page 36, line 4
Although the RPE-LTP algorithm is not an ITU-T standard, there is a C 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 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 ITU-T STL. The STL implementation is an adaptation of the TU Berlin
version. 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 PCMU, PCMA
An implementation of these algorithm is available as part of the An implementation of these algorithm is available as part of the
ITU-T STL, described above. Code to convert between linear and mu-law ITU-T STL, described above. Code to convert between linear and mu-law
companded data is also available in [11]. companded data is also available in [11].
Table of Contents Table of Contents
1 Introduction ........................................ 2 1 Introduction ........................................ 3
1.1 Terminology ......................................... 3 1.1 Terminology ......................................... 3
2 RTP and RTCP Packet Forms and Protocol Behavior ..... 3 2 RTP and RTCP Packet Forms and Protocol Behavior ..... 3
3 Registering Additional Encodings with IANA .......... 6 3 IANA Considerations ................................. 6
4 Audio ............................................... 7 3.1 Registering Additional Encodings .................... 6
4.1 Encoding-Independent Rules .......................... 7 4 Audio ............................................... 8
4.1 Encoding-Independent Rules .......................... 8
4.2 Operating Recommendations ........................... 9 4.2 Operating Recommendations ........................... 9
4.3 Guidelines for Sample-Based Audio Encodings ......... 9 4.3 Guidelines for Sample-Based Audio Encodings ......... 9
4.4 Guidelines for Frame-Based Audio Encodings .......... 10 4.4 Guidelines for Frame-Based Audio Encodings .......... 10
4.5 Audio Encodings ..................................... 10 4.5 Audio Encodings ..................................... 11
4.5.1 1016 ................................................ 11 4.5.1 1016 ................................................ 12
4.5.2 CN .................................................. 11 4.5.2 DVI4 ................................................ 12
4.5.3 DVI4 ................................................ 12 4.5.3 G722 ................................................ 13
4.5.4 G722 ................................................ 13 4.5.4 G723 ................................................ 13
4.5.5 G723 ................................................ 14 4.5.5 G726-32 ............................................. 14
4.5.6 G726-32 ............................................. 14 4.5.6 G728 ................................................ 14
4.5.7 G728 ................................................ 15 4.5.7 G729 ................................................ 15
4.5.8 G729 ................................................ 16 4.5.8 GSM ................................................. 17
4.5.9 GSM ................................................. 17 4.5.8.1 General Packaging Issues ............................ 17
4.5.9.1 General Packaging Issues ............................ 18 4.5.8.2 GSM variable names and numbers ...................... 17
4.5.9.2 GSM variable names and numbers ...................... 18 4.5.9 GSM-HR .............................................. 19
4.5.10 GSM-HR .............................................. 18 4.5.10 GSM-EFR ............................................. 20
4.5.11 GSM-EFR ............................................. 18 4.5.11 L8 .................................................. 20
4.5.12 L8 .................................................. 20 4.5.12 L16 ................................................. 20
4.5.13 L16 ................................................. 20 4.5.13 LPC ................................................. 20
4.5.14 LPC ................................................. 21 4.5.14 MPA ................................................. 20
4.5.15 MPA ................................................. 21 4.5.15 PCMA and PCMU ....................................... 21
4.5.16 PCMA and PCMU ....................................... 21 4.5.16 QCELP ............................................... 21
4.5.17 QCELP ............................................... 21 4.5.17 RED ................................................. 21
4.5.18 RED ................................................. 22 4.5.18 VDVI ................................................ 21
4.5.19 VDVI ................................................ 22 5 Video ............................................... 22
5 Video ............................................... 23
5.1 BT656 ............................................... 23 5.1 BT656 ............................................... 23
5.2 CelB ................................................ 23 5.2 CelB ................................................ 23
5.3 JPEG ................................................ 24 5.3 JPEG ................................................ 23
5.4 H261 ................................................ 24 5.4 H261 ................................................ 23
5.5 H263 ................................................ 24 5.5 H263 ................................................ 23
5.6 H263-1998 ........................................... 24 5.6 H263-1998 ........................................... 23
5.7 MPV ................................................. 24 5.7 MPV ................................................. 24
5.8 MP2T ................................................ 24 5.8 MP2T ................................................ 24
5.9 MP1S ................................................ 25 5.9 MP1S ................................................ 24
5.10 MP2P ................................................ 25 5.10 MP2P ................................................ 24
5.11 BMPEG ............................................... 25 5.11 BMPEG ............................................... 24
5.12 nv .................................................. 25 5.12 nv .................................................. 24
6 Payload Type Definitions ............................ 25 6 Payload Type Definitions ............................ 25
7 RTP over TCP and Similar Byte Stream Protocols ...... 26 7 RTP over TCP and Similar Byte Stream Protocols ...... 26
8 Port Assignment ..................................... 26 8 Port Assignment ..................................... 27
9 Changes from RFC 1890 ............................... 27 9 Changes from RFC 1890 ............................... 28
10 Security Considerations ............................. 30 10 Security Considerations ............................. 29
11 Full Copyright Statement ............................ 30 11 Full Copyright Statement ............................ 30
12 Acknowledgements .................................... 31 12 Acknowledgements .................................... 31
13 Addresses of Authors ................................ 31 13 Addresses of Authors ................................ 31
A Bibliography ........................................ 31 A Bibliography ........................................ 31
 End of changes. 

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