draft-ietf-avt-profile-new-09.txt   draft-ietf-avt-profile-new-10.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-09.txt Columbia U./Packet Design draft-ietf-avt-profile-new-10.txt Columbia U./Packet Design
July 14, 2000 March 2, 2001
Expires: January 14, 2001 Expires: August 2, 2001
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
skipping to change at page 2, line 29 skipping to change at page 2, line 29
YYYY should be filled in with the number of the draft specifying MIME YYYY should be filled in with the number of the draft specifying MIME
registration of RTP payload types as it is submitted for Proposed registration of RTP payload types as it is submitted for Proposed
Standard status. These latter references are intended to be non- Standard status. These latter references are intended to be non-
normative.] normative.]
Readers are directed to Appendix 9, Changes from RFC 1890, for a Readers are directed to Appendix 9, Changes from RFC 1890, for a
listing of the changes that have been made in this draft. The listing of the changes that have been made in this draft. The
changes from RFC 1890 are marked with change bars in the PostScript changes from RFC 1890 are marked with change bars in the PostScript
form of this draft. form of this draft.
The revisions in this draft are intended to be complete for Last The changes in this revision of the draft from the previous one are:
Call. The following open issues from previous drafts have been
addressed: o An paragraph further explaining the requirements for congestion
control was added to Section 2 based on the discussion at IETF
49.
o Packetization of G.726 audio at rates 40, 24 and 16 kb/s is
specified in addition to 32 kb/s.
o The mapping of a user pass-phrase string into an encryption key
was deleted from Section 2 because two interoperable
implementations were not found.
o The specification of a two-byte encapsulation for RTP over TCP
was deleted because two interoperable implementations were not
found.
o The audio payload formats 1016, G723, GSM-HR and GSM-EFR were
removed because two interoperable implementations were not
found.
o The video payload formats H263, BT656, MP2T, MP1S, MP2P and
BMPEG were removed because two interoperable implementations
were not found.
This version of the draft is intended to be complete for Last Call.
The following open issues from previous drafts have been addressed:
o The procedure for registering RTP encoding names as MIME o The procedure for registering RTP encoding names as MIME
subtypes was moved to a separate RFC-to-be that may also serve subtypes was moved to a separate RFC-to-be that may also serve
to specify how (some of) the encodings here may be used with to specify how (some of) the encodings here may be used with
mail and other not-RTP transports. That procedure is not mail and other not-RTP transports. That procedure is not
required to implement this profile, but may be used in those required to implement this profile, but may be used in those
contexts where it is needed. contexts where it is needed.
o This profile follows the suggestion in the RTP spec that RTCP o This profile follows the suggestion in the RTP spec that RTCP
bandwidth may be specified separately from the session bandwidth may be specified separately from the session
skipping to change at page 5, line 25 skipping to change at page 5, line 45
sent every third reporting interval, with NAME sent seven sent every third reporting interval, with NAME sent seven
out of eight times within that slot and the remaining SDES out of eight times within that slot and the remaining SDES
items cyclically taking up the eighth slot, as defined in items cyclically taking up the eighth slot, as defined in
Section 6.2.2 of the RTP specification. In other words, Section 6.2.2 of the RTP specification. In other words,
NAME is sent in RTCP packets 1, 4, 7, 10, 13, 16, 19, NAME is sent in RTCP packets 1, 4, 7, 10, 13, 16, 19,
while, say, EMAIL is used in RTCP packet 22. while, say, EMAIL is used in RTCP packet 22.
Security: The RTP default security services are also the default Security: The RTP default security services are also the default
under this profile. under this profile.
String-to-key mapping: A user-provided string ("pass phrase") is String-to-key mapping: No mapping is specified by this profile.
hashed with the MD5 algorithm to a 16-octet digest. An n-
bit key is extracted from the digest by taking the first n
bits 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 RECOMMENDED that pass phrases be 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 `white space' to be the ASCII space,
formfeed, newline, carriage 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.
Congestion: RTP and this profile may be used in the context of Congestion: RTP and this profile may be used in the context of
enhanced network service, for example, through Integrated enhanced network service, for example, through Integrated
Services (RFC 1633) [3] or Differentiated Services (RFC Services (RFC 1633) [3] or Differentiated Services (RFC
2475) [4], or they may be used with best effort service. 2475) [4], or they may be used with best effort service.
If enhanced service is being used, RTP receivers SHOULD If enhanced service is being used, RTP receivers SHOULD
monitor packet loss to ensure that the service that was monitor packet loss to ensure that the service that was
requested is actually being delivered. If it is not, then requested is actually being delivered. If it is not, then
they SHOULD assume that they are receiving best-effort they SHOULD assume that they are receiving best-effort
service and behave accordingly. service and behave accordingly.
If best-effort service is being used, RTP receivers SHOULD If best-effort service is being used, RTP receivers SHOULD
monitor packet loss to ensure that the packet loss rate is monitor packet loss to ensure that the packet loss rate is
within acceptable parameters. Packet loss is considered within acceptable parameters. Packet loss is considered
acceptable if a TCP flow across the same network path and acceptable if a TCP flow across the same network path and
experiencing the same network conditions would achieve an experiencing the same network conditions would achieve an
average throughput that is not less the RTP flow is average throughput, measured on a reasonable timescale,
achieving. This condition can be satisfied by implementing that is not less the RTP flow is achieving. This condition
congestion control mechanisms to adapt the transmission can be satisfied by implementing congestion control
rate (or the number of layers subscribed for a layered mechanisms to adapt the transmission rate (or the number of
multicast session), or by arranging for a receiver to leave layers subscribed for a layered multicast session), or by
the session if the loss rate is unacceptably high. arranging for a receiver to leave the session if the loss
rate is unacceptably high.
The comparison to TCP cannot be specified exactly, but is
intended as an "order-of-magnitude" comparison in timescale
and throughput. The timescale on which TCP throughput is
measured is the round-trip time of the connection. In
essence, this requirement states that it is not acceptable
to deploy an application (using RTP or any other transport
protocol) on the best-effort Internet which consumes
bandwidth arbitrarily and does not compete fairly with TCP
within an order of magnitude.
Underlying protocol: The profile specifies the use of RTP over 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: This profile leaves to applications the
specification of RTP encapsulation in protocols other
than UDP.
3 IANA Considerations 3 IANA Considerations
The RTP specification establishes a registry of profile names for use The RTP specification establishes a registry of profile names for use
by higher-level control protocols, such as the Session Description by higher-level control protocols, such as the Session Description
Protocol (SDP), RFC 2327 [5], to refer to transport methods. This Protocol (SDP), RFC 2327 [5], to refer to transport methods. This
profile registers the name "RTP/AVP". profile registers the name "RTP/AVP".
3.1 Registering Additional Encodings 3.1 Registering Additional Encodings
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4.5 Audio Encodings 4.5 Audio Encodings
The characteristics of the audio encodings described in this document The characteristics of the audio encodings described in this document
are shown in Table 1; they are listed in order of their payload type 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 in Table 4. While most audio codecs are only specified for a fixed
sampling rate, some sample-based algorithms (indicated by an entry of sampling rate, some sample-based algorithms (indicated by an entry of
"var." in the sampling rate column of Table 1) may be used with "var." in the sampling rate column of Table 1) may be used with
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
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 G726-40 sample 5 8,000 20
G726-32 sample 4 8,000 20 G726-32 sample 4 8,000 20
G726-24 sample 3 8,000 20
G726-16 sample 2 8,000 20
G728 frame N/A 8,000 2.5 20 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
G729D frame N/A 8,000 10 20 G729D frame N/A 8,000 10 20
G729E frame N/A 8,000 10 20 G729E 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-EFR frame N/A 8,000 20 20
L8 sample 8 var. 20 L8 sample 8 var. 20
L16 sample 16 var. 20 L16 sample 16 var. 20
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)
different sampling rates, resulting in different coded bit rates. different sampling rates, resulting in different coded bit rates.
When used with a sampling rate other than that for which a static 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 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 DVI4
Encoding 1016 is a frame based encoding using code-excited linear
prediction (CELP) and is specified in Federal Standard FED-STD 1016
[9,10,11,12].
4.5.2 DVI4
DVI4 is specified, with pseudo-code, in [13] as the IMA ADPCM wave DVI4 is specified, with pseudo-code, in [9] 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.
o IMA ADPCM blocks contain an odd number of samples, since the o IMA ADPCM blocks contain an odd number of samples, since the
first sample of a block is contained just in the header first sample of a block is contained just in the header
skipping to change at page 14, line 5 skipping to change at page 14, line 5
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
4.5.3 G722 4.5.2 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.
4.5.4 G723 4.5.3 G726-40, G726-32, G726-24, and G726-16
G723 is specified in ITU Recommendation G.723.1, "Dual-rate speech
coder for multimedia communications transmitting at 5.3 and 6.3
kbit/s". The G.723.1 5.3/6.3 kbit/s codec was defined by the ITU-T as
a mandatory codec for ITU-T H.324 GSTN videophone terminal
applications. The algorithm has a floating point specification in
Annex B to G.723.1, a silence compression algorithm in Annex A to
G.723.1 and an encoded signal bit-error sensitivity specification in
G.723.1 Annex C.
This Recommendation specifies a coded representation that can be used
for compressing the speech signal component of multi-media services
at a very low bit rate. Audio is encoded in 30 ms frames, with an
additional delay of 7.5 ms due to look-ahead. A G.723.1 frame can be
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. This coder was optimized to represent speech
with near-toll quality at the above rates using a limited amount of
complexity.
The packing of the encoded bit stream into octets and the
transmission order of the octets is specified in Rec. G.723.1 and is
the same as that produced by the G.723 C code reference
implementation. For the 6.3 kb/s data rate, this packing is
illustrated as follows, where the header (HDR) bits are always "0 0"
as shown in Fig. 1 to indicate operation at 6.3 kb/s, and the Z bit
is always set to zero. The diagrams show the bit packing in "network
byte order," also known as big-endian order. The bits of each 32-bit
word are numbered 0 to 31, with the most significant bit on the left
and numbered 0. The octets (bytes) of each word are transmitted most
significant octet first. The bits of each data field are numbered in
the order of the bit stream representation of the encoding (least
significant bit first). The vertical bars indicate the boundaries
between field fragments.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| LPC |HDR| LPC | LPC | ACL0 |LPC|
| | | | | | |
|0 0 0 0 0 0|0 0|1 1 1 1 0 0 0 0|2 2 1 1 1 1 1 1|0 0 0 0 0 0|2 2|
|5 4 3 2 1 0| |3 2 1 0 9 8 7 6|1 0 9 8 7 6 5 4|5 4 3 2 1 0|3 2|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ACL2 |ACL|A| GAIN0 |ACL|ACL| GAIN0 | GAIN1 |
| | 1 |C| | 3 | 2 | | |
|0 0 0 0 0|0 0|0|0 0 0 0|0 0|0 0|1 1 0 0 0 0 0 0|0 0 0 0 0 0 0 0|
|4 3 2 1 0|1 0|6|3 2 1 0|1 0|6 5|1 0 9 8 7 6 5 4|7 6 5 4 3 2 1 0|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| GAIN2 | GAIN1 | GAIN2 | GAIN3 | GRID | GAIN3 |
| | | | | | |
|0 0 0 0|1 1 0 0|1 1 0 0 0 0 0 0|0 0 0 0 0 0 0 0|0 0 0 0|1 1 0 0|
|3 2 1 0|1 0 9 8|1 0 9 8 7 6 5 4|7 6 5 4 3 2 1 0|3 2 1 0|1 0 9 8|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| MSBPOS |Z|POS| MSBPOS | POS0 |POS| POS0 |
| | | 0 | | | 1 | |
|0 0 0 0 0 0 0|0|0 0|1 1 1 0 0 0|0 0 0 0 0 0 0 0|0 0|1 1 1 1 1 1|
|6 5 4 3 2 1 0| |1 0|2 1 0 9 8 7|9 8 7 6 5 4 3 2|1 0|5 4 3 2 1 0|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| POS1 | POS2 | POS1 | POS2 | POS3 | POS2 |
| | | | | | |
|0 0 0 0 0 0 0 0|0 0 0 0|1 1 1 1|1 1 0 0 0 0 0 0|0 0 0 0|1 1 1 1|
|9 8 7 6 5 4 3 2|3 2 1 0|3 2 1 0|1 0 9 8 7 6 5 4|3 2 1 0|5 4 3 2|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| POS3 | PSIG0 |POS|PSIG2| PSIG1 | PSIG3 |PSIG2|
| | | 3 | | | | |
|1 1 0 0 0 0 0 0|0 0 0 0 0 0|1 1|0 0 0|0 0 0 0 0|0 0 0 0 0|0 0 0|
|1 0 9 8 7 6 5 4|5 4 3 2 1 0|3 2|2 1 0|4 3 2 1 0|4 3 2 1 0|5 4 3|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 1: G.723 (6.3 kb/s) bit packing
For the 5.3 kb/s data rate, the header (HDR) bits are always "0 1",
as shown in Fig. 2, to indicate operation at 5.3 kb/s.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| LPC |HDR| LPC | LPC | ACL0 |LPC|
| | | | | | |
|0 0 0 0 0 0|0 1|1 1 1 1 0 0 0 0|2 2 1 1 1 1 1 1|0 0 0 0 0 0|2 2|
|5 4 3 2 1 0| |3 2 1 0 9 8 7 6|1 0 9 8 7 6 5 4|5 4 3 2 1 0|3 2|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ACL2 |ACL|A| GAIN0 |ACL|ACL| GAIN0 | GAIN1 |
| | 1 |C| | 3 | 2 | | |
|0 0 0 0 0|0 0|0|0 0 0 0|0 0|0 0|1 1 0 0 0 0 0 0|0 0 0 0 0 0 0 0|
|4 3 2 1 0|1 0|6|3 2 1 0|1 0|6 5|1 0 9 8 7 6 5 4|7 6 5 4 3 2 1 0|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| GAIN2 | GAIN1 | GAIN2 | GAIN3 | GRID | GAIN3 |
| | | | | | |
|0 0 0 0|1 1 0 0|1 1 0 0 0 0 0 0|0 0 0 0 0 0 0 0|0 0 0 0|1 1 0 0|
|3 2 1 0|1 0 9 8|1 0 9 8 7 6 5 4|7 6 5 4 3 2 1 0|4 3 2 1|1 0 9 8|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| POS0 | POS1 | POS0 | POS1 | POS2 |
| | | | | |
|0 0 0 0 0 0 0 0|0 0 0 0|1 1 0 0|1 1 0 0 0 0 0 0|0 0 0 0 0 0 0 0|
|7 6 5 4 3 2 1 0|3 2 1 0|1 0 9 8|1 0 9 8 7 6 5 4|7 6 5 4 3 2 1 0|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| POS3 | POS2 | POS3 | PSIG1 | PSIG0 | PSIG3 | PSIG2 |
| | | | | | | |
|0 0 0 0|1 1 0 0|1 1 0 0 0 0 0 0|0 0 0 0|0 0 0 0|0 0 0 0|0 0 0 0|
|3 2 1 0|1 0 9 8|1 0 9 8 7 6 5 4|3 2 1 0|3 2 1 0|3 2 1 0|3 2 1 0|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 2: G.723 (5.3 kb/s) bit packing ITU-T Recommendation G.726 describes, among others, the algorithm
recommended for conversion of a single 64 kbit/s A-law or mu-law
PCM channel encoded at 8000 samples/sec to and from a 40, 32, 24,
or 16 kbit/s channel. The conversion is applied to the PCM stream
using an Adaptive Differential Pulse Code Modulation (ADPCM)
transcoding technique. The ADPCM representation consists of a
series of codewords with a one-to-one correspondance to the samples
in the PCM stream. The G726 data rates of 40, 32, 24, and 16
kbit/s have codewords of 5, 4, 3, and 2 bits respectively.
The packing of G.723.1 SID (silence) frames, which are indicated by The 16 and 24 kbit/s encodings do not provide toll quality speech.
the header (HDR) bits having the pattern "1 0", is depicted in Fig. They are designed for used in overloaded Digital Circuit
3. Multiplication Equipment (DCME). ITU-T G.726 recommends that the
16 and 24 kbit/s encodings should be alternated with higher data
rate encodings to provide an average sample size of between 3.5 and
3.7 bits per sample.
0 1 2 3 The encodings of G.726 are here denoted as G726-40, G726-32,
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 G726-24, and G726-16. Prior to 1990, G721 described the 32 kbit/s
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ ADPCM encoding, and G723 described the 40, 32, and 16 kbit/s
| LPC |HDR| LPC | LPC | GAIN |LPC| encodings. Thus, G726-32 designates the same algorithm as G721 in
| | | | | | | RFC 1890.
|0 0 0 0 0 0|1 0|1 1 1 1 0 0 0 0|2 2 1 1 1 1 1 1|0 0 0 0 0 0|2 2|
|5 4 3 2 1 0| |3 2 1 0 9 8 7 6|1 0 9 8 7 6 5 4|5 4 3 2 1 0|3 2|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 3: G.723 SID mode bit packing A stream of G726 codewords contains no information on the encoding
being used, therefore transitions between G726 encoding types is
not permitted within a sequence of packed codewords. Applications
MUST determine the encoding type of packed codewords from the RTP
payload identifier.
4.5.5 G726-32 No payload-specific header information SHALL be included as part
of the audio data. A stream of G726 codewords MUST be packed into
octets as follows: the first codeword is placed into the first
octet such that the least significant bit of the codeword aligns
with the least significant bit in the octet, the second codeword
is then packed so that its least significant bit coincides with
the least significant unoccupied bit in the octet. When a
complete codeword cannot be placed into an octet, the bits
overlapping the octet boundary are placed into the least
significant bits of the next octet. Packing MUST end with a
completely packed final octet. The number of codewords packed
will therefore be a multiple of 8, 2, 8, and 4 for G726-40,
G726-32, G726-24, and G726-16 respectively. An examples of the
packing scheme for G726-32 codewords is as shown:
ITU-T Recommendation G.726 describes, among others, the algorithm 0 1
recommended for conversion of a single 64 kbit/s A-law or mu-law PCM 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
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 |B B B B|A A A A|D D D D|C C C C| ...
Differential Pulse Code Modulation (ADPCM) transcoding technique. |0 1 2 3|0 1 2 3|0 1 2 3|0 1 2 3|
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).
Packetization is specified here only for the 32 kb/s encoding which
is labeled G726-32.
Note: In 1990, ITU-T Recommendation G.721 was merged with An example of the packing scheme for G726-24 codewords is:
Recommendation G.723 into ITU-T Recommendation G.726. Thus, G726-32
designates the same algorithm as G721 in RFC 1890.
No payload-specific header information SHALL be included as part of 0 1 2
the audio data. The 4-bit code words of the G726-32 encoding MUST be 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
packed into octets as follows: the first code word is placed in the +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
four least significant bits of the first octet, with the least |C C|B B B|A A A|F|E E E|D D D|C|H H H|G G G|F F| ...
significant bit of the code word in the least significant bit of the |1 2|0 1 2|0 1 2|2|0 1 2|0 1 2|0|0 1 2|0 1 2|0 1|
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. The number of samples per packet
MUST be even because there is no means to indicate a partially filled
last octet.
4.5.6 G728 4.5.4 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
skipping to change at page 19, line 31 skipping to change at page 16, line 36
<---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.
4.5.7 G729 4.5.5 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
reduction [14]. See the next Section (4.5.8) for other data rates reduction [10]. See the next Section (4.5.6) for other data rates
added in later G.729 Annexes. For all data rates, the sampling added in later G.729 Annexes. For all data rates, the sampling
frequency (and RTP timestamp clock rate) is 8000 Hz. frequency (and RTP timestamp clock rate) is 8000 Hz.
A voice activity detector (VAD) and comfort noise generator (CNG) A voice activity detector (VAD) and comfort noise generator (CNG)
algorithm in Annex B of G.729 is RECOMMENDED for digital simultaneous algorithm in Annex B of G.729 is RECOMMENDED for digital simultaneous
voice and data applications and can be used in conjunction with G.729 voice and data applications and can be used in conjunction with G.729
or G.729 Annex A. A G.729 or G.729 Annex A frame contains 10 octets, or G.729 Annex A. A G.729 or G.729 Annex A frame contains 10 octets,
while the G.729 Annex B comfort noise frame occupies 2 octets. while the G.729 Annex B comfort noise frame occupies 2 octets.
A G729 RTP packet may consist of zero or more G.729 or G.729 Annex A A G729 RTP packet may consist of zero or more G.729 or G.729 Annex A
skipping to change at page 20, line 45 skipping to change at page 18, line 4
| |0 1 2 3 4 5 6|0 1 2 3 4|0 1 2 3 4|0 1 2 3 4 5 6 7| |0 1 2 3 4| | |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 | | C1 | S1 | GA1 | GB1 | P2 | C2 |
| 1 1 1| | | | | | | 1 1 1| | | | | |
|5 6 7 8 9 0 1 2|0 1 2 3|0 1 2|0 1 2 3|0 1 2 3 4|0 1 2 3 4 5 6 7| |5 6 7 8 9 0 1 2|0 1 2 3|0 1 2|0 1 2 3|0 1 2 3 4|0 1 2 3 4 5 6 7|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| C2 | S2 | GA2 | GB2 | | C2 | S2 | GA2 | GB2 |
| 1 1 1| | | | | 1 1 1| | | |
|8 9 0 1 2|0 1 2 3|0 1 2|0 1 2 3| |8 9 0 1 2|0 1 2 3|0 1 2|0 1 2 3|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The packing of the G.729 Annex B comfort noise frame is as follows:
Figure 4: G.729 and G.729A bit packing
The packing of the G.729 Annex B comfort noise frame is shown in Fig.
5.
0 1 0 1
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|L| LSF1 | LSF2 | GAIN |R| |L| LSF1 | LSF2 | GAIN |R|
|S| | | |E| |S| | | |E|
|F| | | |S| |F| | | |S|
|0|0 1 2 3 4|0 1 2 3|0 1 2 3 4|V| RESV = Reserved (zero) |0|0 1 2 3 4|0 1 2 3|0 1 2 3 4|V| RESV = Reserved (zero)
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 5: G.729 Annex B bit packing 4.5.6 G729D and G729E
4.5.8 G729D and G729E
Annexes D and E to ITU-T Recommendation G.729 provide additional data Annexes D and E to ITU-T Recommendation G.729 provide additional data
rates. Because the data rate is not signaled in the bitstream, the rates. Because the data rate is not signaled in the bitstream, the
different data rates are given distinct RTP encoding names which are different data rates are given distinct RTP encoding names which are
mapped to distinct payload type numbers. G729D indicates a 6.4 kbit/s mapped to distinct payload type numbers. G729D indicates a 6.4 kbit/s
coding mode (G.729 Annex D, for momentary reduction in channel coding mode (G.729 Annex D, for momentary reduction in channel
capacity), while G729E indicates an 11.8 kbit/s mode (G.729 Annex E, capacity), while G729E indicates an 11.8 kbit/s mode (G.729 Annex E,
for improved performance with a wide range of narrow-band input for improved performance with a wide range of narrow-band input
signals, e.g. music and background noise). Annex E has two operating signals, e.g. music and background noise). Annex E has two operating
modes, backward adaptive and forward adaptive, which are signaled by modes, backward adaptive and forward adaptive, which are signaled by
skipping to change at page 22, line 20 skipping to change at page 19, line 20
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|L| L1 | L2 | L3 | P1 | C1 | |L| L1 | L2 | L3 | P1 | 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 5| | |0 1 2 3 4 5 6|0 1 2 3 4|0 1 2 3 4|0 1 2 3 4 5 6 7|0 1 2 3 4 5|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| C1 |S1 | GA1 | GB1 | P2 | C2 |S2 | GA2 | GB2 | | C1 |S1 | GA1 | GB1 | P2 | C2 |S2 | GA2 | GB2 |
| | | | | | | | | | | | | | | | | | | |
|6 7 8|0 1|0 1 2|0 1 2|0 1 2 3|0 1 2 3 4 5 6 7 8|0 1|0 1 2|0 1 2| |6 7 8|0 1|0 1 2|0 1 2|0 1 2 3|0 1 2 3 4 5 6 7 8|0 1|0 1 2|0 1 2|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 6: G.729 Annex D bit packing
The net bit rate for the G.729 Annex E algorithm is 11.8 kbit/s and a The net bit rate for the G.729 Annex E algorithm is 11.8 kbit/s and a
total of 118 bits are used. Two bits are appended as "don't care" total of 118 bits are used. Two bits are appended as "don't care"
bits to complete an integer number of octets for the frame. For bits to complete an integer number of octets for the frame. For
G729E, the bits of a data frame are formatted as shown in the next G729E, the bits of a data frame are formatted as shown in the next
two diagrams (cf. Table E.1/G.729). The fields for the G729E forward two diagrams (cf. Table E.1/G.729). The fields for the G729E forward
adaptive mode are packed as shown in Fig. 7. adaptive mode are packed as follows:
0 1 2 3 0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0 0|L| L1 | L2 | L3 | P1 |P| C0_1| |0 0|L| L1 | L2 | L3 | P1 |P| C0_1|
| |0| | | | |0| | | |0| | | | |0| |
| | |0 1 2 3 4 5 6|0 1 2 3 4|0 1 2 3 4|0 1 2 3 4 5 6 7| |0 1 2| | | |0 1 2 3 4 5 6|0 1 2 3 4|0 1 2 3 4|0 1 2 3 4 5 6 7| |0 1 2|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| | C1_1 | C2_1 | C3_1 | C4_1 | | | C1_1 | C2_1 | C3_1 | C4_1 |
| | | | | | | | | | | |
|3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5 6| |3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| GA1 | GB1 | P2 | C0_2 | C1_2 | C2_2 | | GA1 | GB1 | P2 | C0_2 | C1_2 | C2_2 |
| | | | | | | | | | | | | |
|0 1 2|0 1 2 3|0 1 2 3 4|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5| |0 1 2|0 1 2 3|0 1 2 3 4|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2 3 4 5|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| | C3_2 | C4_2 | GA2 | GB2 |DC | | | C3_2 | C4_2 | GA2 | GB2 |DC |
| | | | | | | | | | | | | |
|6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2|0 1 2 3|0 1| |6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2|0 1 2 3|0 1|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 7: G.729 Annex E (forward adaptive mode) bit packing
The fields for the G729E backward adaptive mode are packed as shown The fields for the G729E backward adaptive mode are packed as shown
in Fig. 8. in Fig. 8.
0 1 2 3 0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1 1| P1 |P| C0_1 | C1_1 | |1 1| P1 |P| C0_1 | C1_1 |
| | |0| 1 1 1| | | | |0| 1 1 1| |
| |0 1 2 3 4 5 6 7|0|0 1 2 3 4 5 6 7 8 9 0 1 2|0 1 2 3 4 5 6 7| | |0 1 2 3 4 5 6 7|0|0 1 2 3 4 5 6 7 8 9 0 1 2|0 1 2 3 4 5 6 7|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
skipping to change at page 24, line 25 skipping to change at page 21, line 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| | C0_2 | C1_2 | C2_2 | | | C0_2 | C1_2 | C2_2 |
| | 1 1 1| | | | | 1 1 1| | |
|2 3 4|0 1 2 3 4 5 6 7 8 9 0 1 2|0 1 2 3 4 5 6 7 8 9|0 1 2 3 4 5| |2 3 4|0 1 2 3 4 5 6 7 8 9 0 1 2|0 1 2 3 4 5 6 7 8 9|0 1 2 3 4 5|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| | C3_2 | C4_2 | GA2 | GB2 |DC | | | C3_2 | C4_2 | GA2 | GB2 |DC |
| | | | | | | | | | | | | |
|6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2|0 1 2 3|0 1| |6|0 1 2 3 4 5 6|0 1 2 3 4 5 6|0 1 2|0 1 2 3|0 1|
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 8: G.729 Annex E (backward adaptive mode) bit packing 4.5.7 GSM
4.5.9 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 [15,16,17]. The text of the standard can be obtained rate of 13 kb/s [15,16,17]. 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 4.5.7.1 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
skipping to change at page 25, line 18 skipping to change at page 21, line 43
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 4.5.7.2 GSM variable names and numbers
In the RTP encoding we have the bit pattern described in Table 3, 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 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 significant bit, and the bits of every octet are numbered from 0 to 7
from most to least significant. from most to least significant.
4.5.10 GSM-HR 4.5.8 L8
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
similar to that specified here for the original GSM 06.10 codec. The
packing is specified in ETSI Technical Specification TS 101 318.
4.5.12 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.
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
skipping to change at page 26, line 48 skipping to change at page 22, line 48
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
4.5.13 L16 4.5.9 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
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
skipping to change at page 27, line 46 skipping to change at page 23, line 46
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
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 4.5.10 LPC
LPC designates an experimental linear predictive encoding contributed LPC designates an experimental linear predictive encoding contributed
by Ron Frederick, which is based on an implementation written by Ron by Ron Frederick, which is based on an implementation written by Ron
Zuckerman posted to the Usenet group comp.dsp on June 26, 1992. The Zuckerman posted to the Usenet group comp.dsp on June 26, 1992. The
codec generates 14 octets for every frame. The framesize is set to 20 codec generates 14 octets for every frame. The framesize is set to 20
ms, resulting in a bit rate of 5,600 b/s. ms, resulting in a bit rate of 5,600 b/s.
4.5.15 MPA 4.5.11 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 [18]. and 13818-3. The encapsulation is specified in RFC 2250 [14].
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 4.5.12 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 [19]. Each G.711 octet SHALL be octet- given by Jayant and Noll [15]. Each G.711 octet SHALL be octet-
aligned in an RTP packet. The sign bit of each G.711 octet SHALL 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 MUST 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 4.5.13 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 [20]. audio codec is described in [16].
4.5.18 RED 4.5.14 RED
The redundant audio payload format "RED" is specified by RFC 2198 The redundant audio payload format "RED" is specified by RFC 2198
[21]. It defines a means by which multiple redundant copies of an [17]. 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 4.5.15 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 30, line 30 skipping to change at page 26, line 30
If a video image occupies more than one packet, the timestamp is the If a video image occupies more than one packet, the timestamp is the
same on all of those packets. Packets from different video images are same on all of those packets. Packets from different video images are
distinguished by their different timestamps. distinguished by their different timestamps.
Most of these video encodings also specify that the marker bit of the Most of these video encodings also specify that the marker bit of the
RTP header SHOULD be set to one in the last packet of a video frame RTP header SHOULD be set to one in the last packet of a video frame
and otherwise set to zero. Thus, it is not necessary to wait for a and otherwise set to zero. Thus, it is not necessary to wait for a
following packet with a different timestamp to detect that a new following packet with a different timestamp to detect that a new
frame should be displayed. frame should be displayed.
5.1 BT656 5.1 CelB
The encoding is specified in ITU-R Recommendation BT.656-3,
"Interfaces for Digital Component Video Signals in 525-Line and 625-
Line Television Systems operating at the 4:2:2 Level of
Recommendation ITU-R BT.601 (Part A)". The packetization and RTP-
specific properties are described in RFC 2431 [22].
5.2 CelB
The CELL-B encoding is a proprietary encoding proposed by Sun The CELL-B encoding is a proprietary encoding proposed by Sun
Microsystems. The byte stream format is described in RFC 2029 [23]. Microsystems. The byte stream format is described in RFC 2029 [18].
5.3 JPEG 5.2 JPEG
The encoding is specified in ISO Standards 10918-1 and 10918-2. The The encoding is specified in ISO Standards 10918-1 and 10918-2. The
RTP payload format is as specified in RFC 2435 [24]. RTP payload format is as specified in RFC 2435 [19].
5.4 H261 5.3 H261
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 [25]. RTP-specific properties are described in RFC 2032 [20].
5.5 H263
The encoding is specified in the 1996 version of ITU-T Recommendation
H.263, "Video coding for low bit rate communication". The
packetization and RTP-specific properties are described in RFC 2190
[26]. The H263-1998 payload format is RECOMMENDED over this one for
use by new implementations.
5.6 H263-1998 5.4 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
[27]. Because the 1998 version of H.263 is a superset of the 1996 [21]. 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
described in RFC 2429. described in RFC 2429.
5.7 MPV 5.5 MPV
MPV designates the use of MPEG-1 and MPEG-2 video encoding elementary MPV designates the use of MPEG-1 and MPEG-2 video encoding elementary
streams as specified in ISO Standards ISO/IEC 11172 and 13818-2, streams as specified in ISO Standards ISO/IEC 11172 and 13818-2,
respectively. The RTP payload format is as specified in RFC 2250 respectively. The RTP payload format is as specified in RFC 2250
[18], Section 3. [14], Section 3.
5.8 MP2T
MP2T designates the use of MPEG-2 transport streams, for either audio
or video. The RTP payoad format is described in RFC 2250 [18],
Section 2.
5.9 MP1S
MP1S designates an MPEG-1 systems stream, encapsulated according to
RFC 2250 [18].
5.10 MP2P
MP2P designates an MPEG-2 program stream, encapsulated according to
RFC 2250 [18].
5.11 BMPEG
BMPEG designates an experimental payload format for MPEG-1 and MPEG-2
which specifies bundled (multiplexed) transport of audio and video
elementary streams in one RTP stream as an alternative to the MP1S
and MP2P formats. The packetization is described in RFC 2343 [28].
5.12 nv 5.8 nv
The encoding is implemented in the program `nv', version 4, developed The encoding is implemented in the program `nv', version 4, developed
at Xerox PARC by Ron Frederick. Further information is available from at Xerox PARC by Ron Frederick. Further information is available from
the author: the author:
Ron Frederick Ron Frederick
Entera, Inc. Entera, Inc.
40971 Encyclopedia Circle 40971 Encyclopedia Circle
Fremont, CA 94538 Fremont, CA 94538
United States United States
skipping to change at page 33, line 17 skipping to change at page 29, line 9
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.
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 reserved A
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 reserved A
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 reserved 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 reserved A 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 G726-40 A 8000 1
dyn G726-24 A 8000 1
dyn G726-16 A 8000 1
dyn G729D A 8000 1 dyn G729D A 8000 1
dyn G729E A 8000 1 dyn G729E A 8000 1
dyn GSM-HR A 8000 1
dyn GSM-EFR A 8000 1
dyn L8 A var. var. dyn L8 A var. var.
dyn RED A (see text) dyn RED A (see text)
dyn VDVI A var. 1 dyn VDVI A var. 1
Table 4: Payload types (PT) for audio encodings Table 4: Payload types (PT) for audio encodings
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
31 H261 V 90000 31 H261 V 90000
32 MPV V 90000 32 MPV V 90000
33 MP2T AV 90000 33 reserved V
34 H263 V 90000 34 reserved V
35-71 unassigned ? 35-71 unassigned ?
72-76 reserved N/A N/A 72-76 reserved N/A N/A
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 MP2P 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
7 RTP over TCP and Similar Byte Stream Protocols 7 RTP over TCP and Similar Byte Stream Protocols
Under special circumstances, it may be necessary to carry RTP in Under special circumstances, it may be necessary to carry RTP in
protocols offering a byte stream abstraction, such as TCP, possibly protocols offering a byte stream abstraction, such as TCP, possibly
multiplexed with other data. If the application does not define its multiplexed with other data. The application MUST define its own
own method of delineating RTP and RTCP packets, it SHOULD prefix each method of delineating RTP and RTCP packets (RTSP [22] provides an
packet with a two-octet length field in network order (most example of such an encapsulation specification.)
significant octet first).
(Note: RTSP [29] provides its own encapsulation and does not need an
extra length indication.)
8 Port Assignment 8 Port Assignment
As specified in the RTP protocol definition, RTP data SHOULD be As specified in the RTP protocol definition, RTP data SHOULD be
carried on an even UDP or TCP port number and the corresponding RTCP carried on an even UDP port number and the corresponding RTCP
packets SHOULD be carried on the next higher (odd) port number. packets SHOULD be carried on the next higher (odd) port number.
Applications operating under this profile MAY use any such UDP or TCP Applications operating under this profile MAY use any such UDP port
port pair. For example, the port pair MAY be allocated randomly by a pair. For example, the port pair MAY be allocated randomly by a
session management program. A single fixed port number pair cannot be session management program. A single fixed port number pair cannot be
required because multiple applications using this profile are likely required because multiple applications using this profile are likely
to run on the same host, and there are some operating systems that do 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 not allow multiple processes to use the same UDP port with different
multicast addresses. multicast addresses.
However, port numbers 5004 and 5005 have been registered for use with However, port numbers 5004 and 5005 have been registered for use with
this profile for those applications that choose to use them as the this profile for those applications that choose to use them as the
default pair. Applications that operate under multiple profiles MAY default pair. Applications that operate under multiple profiles MAY
use this port pair as an indication to select this profile if they use this port pair as an indication to select this profile if they
skipping to change at page 35, line 25 skipping to change at page 31, line 20
Applications need not have a default and MAY require that the port Applications need not have a default and MAY require that the port
pair be explicitly specified. The particular port numbers were chosen pair be explicitly specified. The particular port numbers were chosen
to lie in the range above 5000 to accommodate port number allocation to lie in the range above 5000 to accommodate port number allocation
practice within some versions of the Unix operating system, where practice within some versions of the Unix operating system, where
port numbers below 1024 can only be used by privileged processes and port numbers below 1024 can only be used by privileged processes and
port numbers between 1024 and 5000 are automatically assigned by the port numbers between 1024 and 5000 are automatically assigned by the
operating system. operating system.
9 Changes from RFC 1890 9 Changes from RFC 1890
This RFC revises RFC 1890. It is fully backwards-compatible with RFC This RFC revises RFC 1890. It is mostly 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 The mapping of a user pass-phrase string into an encryption
key was deleted from Section 2 because two interoperable
implementations were not found.
o The payload formats for 1016 audio and MP2T video were removed
and their static payload type assignments 1 and 33 were marked
"reserved" because two interoperable implementations were not
found.
o Additional payload formats and/or expanded descriptions were o Additional payload formats and/or expanded descriptions were
included for G722, G723, G726, G728, G729, GSM, GSM-HR, GSM- included for G722, G726, G728, G729, GSM, QCELP, RED, VDVI,
EFR, QCELP, RED, VDVI, BT656, H263, H263-1998, MP1S, MP2P and and H263-1998.
BMPEG.
o Static payload types 4, 12, 16, 17, 18 and 34 were added, and o Static payload types 12, 16, 17 and 18 were added, and 13 and
13 and 19 were reserved. 19 were reserved.
o Requirements for congestion control were added in Section 2. o Requirements for congestion control were added in Section 2.
o A new Section "IANA Considerations" was added to specify the o A new Section "IANA Considerations" was added to specify the
regstration of the name for this profile and to establish a regstration of the name for this profile and to establish a
new policy that no additional registration of static payload new policy that no additional registration of static payload
types for this profile will be made beyond those included in types for this profile will be made beyond those included in
Tables 4 and 5, but that additional encoding names may be Tables 4 and 5, but that additional encoding names may be
registered as MIME subtypes for binding to dynamic payload registered as MIME subtypes for binding to dynamic payload
types. types.
skipping to change at page 36, line 14 skipping to change at page 32, line 17
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
bandwidth may be specified separately from the session bandwidth may be specified separately from the session
bandwidth and separately for active senders and passive bandwidth and separately for active senders and passive
receivers. receivers.
o RFC references are added for payload formats published after o RFC references are added for payload formats published after
RFC 1890. RFC 1890.
o A minimal TCP encapsulation is defined.
o The security considerations and full copyright sections were o The security considerations and full copyright sections were
added. added.
o According to Peter Hoddie of Apple, only pre-1994 Macintosh o According to Peter Hoddie of Apple, only pre-1994 Macintosh
used the 22254.54 rate and none the 11127.27 rate, so the used the 22254.54 rate and none the 11127.27 rate, so the
latter was dropped from the discussion of suggested sampling latter was dropped from the discussion of suggested sampling
frequencies. frequencies.
o Table 1 was corrected to move some values from the "ms/packet" o Table 1 was corrected to move some values from the "ms/packet"
column to the "default ms/packet" column where they belonged. column to the "default ms/packet" column where they belonged.
skipping to change at page 37, line 35 skipping to change at page 33, line 35
into the stream which are complex to decode and cause the receiver to into the stream which are complex to decode and cause the receiver to
be overloaded. However, the encodings described in this profile do be overloaded. However, the encodings described in this profile do
not exhibit any significant non-uniformity. not exhibit any significant non-uniformity.
As with any IP-based protocol, in some circumstances a receiver may As with any IP-based protocol, in some circumstances a receiver may
be overloaded simply by the receipt of too many packets, either be overloaded simply by the receipt of too many packets, either
desired or undesired. Network-layer authentication MAY be used to desired or undesired. Network-layer authentication MAY be used to
discard packets from undesired sources, but the processing cost of discard packets from undesired sources, but the processing cost of
the authentication itself may be too high. In a multicast the authentication itself may be too high. In a multicast
environment, pruning of specific sources may be implemented in future environment, pruning of specific sources may be implemented in future
versions of IGMP [30] and in multicast routing protocols to allow a versions of IGMP [23] and in multicast routing protocols to allow a
receiver to select which sources are allowed to reach it. receiver to select which sources are allowed to reach it.
11 Full Copyright Statement 11 Full Copyright Statement
Copyright (C) The Internet Society (2000). All Rights Reserved. Copyright (C) The Internet Society (2000). All Rights Reserved.
This document and translations of it may be copied and furnished to This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it others, and derivative works that comment on or otherwise explain it
or assist in its implmentation may be prepared, copied, published and or assist in its implmentation may be prepared, copied, published and
distributed, in whole or in part, without restriction of any kind, distributed, in whole or in part, without restriction of any kind,
skipping to change at page 38, line 36 skipping to change at page 34, line 36
Henning Schulzrinne Henning Schulzrinne
Dept. of Computer Science Dept. of Computer Science
Columbia University Columbia University
1214 Amsterdam Avenue 1214 Amsterdam Avenue
New York, NY 10027 New York, NY 10027
USA USA
electronic mail: schulzrinne@cs.columbia.edu electronic mail: schulzrinne@cs.columbia.edu
Stephen L. Casner Stephen L. Casner
Packet Design, Inc. Packet Design
66 Willow Place 2465 Latham Street
Menlo Park, CA 94025 Mountain View, CA 94040
United States United States
electronic mail: casner@acm.org electronic mail: casner@acm.org
A Bibliography A Bibliography
[1] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: A [1] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: A
transport protocol for real-time applications," Internet Draft, transport protocol for real-time applications," Internet Draft,
Internet Engineering Task Force, Feb. 1999 Work in progress, revision Internet Engineering Task Force, Feb. 1999 Work in progress, revision
to RFC 1889. to RFC 1889.
skipping to change at page 39, line 29 skipping to change at page 35, line 29
Internet Draft, Internet Engineering Task Force, Feb. 1999 Work in Internet Draft, Internet Engineering Task Force, Feb. 1999 Work in
progress. progress.
[7] N. Freed, J. Klensin, and J. Postel, "Multipurpose Internet Mail [7] N. Freed, J. Klensin, and J. Postel, "Multipurpose Internet Mail
Extensions (MIME) Part Four: Registration Procedures," RFC 2048, Extensions (MIME) Part Four: Registration Procedures," RFC 2048,
Internet Engineering Task Force, Nov. 1996. Internet Engineering Task Force, Nov. 1996.
[8] Apple Computer, "Audio interchange file format AIFF-C," Aug. [8] Apple Computer, "Audio interchange file format AIFF-C," Aug.
1991. (also ftp://ftp.sgi.com/sgi/aiff-c.9.26.91.ps.Z). 1991. (also ftp://ftp.sgi.com/sgi/aiff-c.9.26.91.ps.Z).
[9] Office of Technology and Standards, "Telecommunications: Analog [9] IMA Digital Audio Focus and Technical Working Groups,
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.
[10] 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.
[11] 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.
[12] 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.
[13] IMA Digital Audio Focus and Technical Working Groups,
"Recommended practices for enhancing digital audio compatibility in "Recommended practices for enhancing digital audio compatibility in
multimedia systems (version 3.00)," tech. rep., Interactive multimedia systems (version 3.00)," tech. rep., Interactive
Multimedia Association, Annapolis, Maryland, Oct. 1992. Multimedia Association, Annapolis, Maryland, Oct. 1992.
[14] D. Deleam and J.-P. Petit, "Real-time implementations of the [10] D. Deleam and J.-P. Petit, "Real-time implementations of the
recent ITU-T low bit rate speech coders on the TI TMS320C54X DSP: recent ITU-T low bit rate speech coders on the TI TMS320C54X DSP:
results, methodology, and applications," in Proc. of International results, methodology, and applications," in Proc. of International
Conference on Signal Processing, Technology, and Applications Conference on Signal Processing, Technology, and Applications
(ICSPAT) , (Boston, Massachusetts), pp. 1656--1660, Oct. 1996. (ICSPAT) , (Boston, Massachusetts), pp. 1656--1660, Oct. 1996.
[15] M. Mouly and M.-B. Pautet, The GSM system for mobile [11] M. Mouly and M.-B. Pautet, The GSM system for mobile
communications Lassay-les-Chateaux, France: Europe Media Duplication, communications Lassay-les-Chateaux, France: Europe Media Duplication,
1993. 1993.
[16] J. Degener, "Digital speech compression," Dr. Dobb's Journal , [12] J. Degener, "Digital speech compression," Dr. Dobb's Journal ,
Dec. 1994. Dec. 1994.
[17] S. M. Redl, M. K. Weber, and M. W. Oliphant, An Introduction to [13] S. M. Redl, M. K. Weber, and M. W. Oliphant, An Introduction to
GSM Boston: Artech House, 1995. GSM Boston: Artech House, 1995.
[18] D. Hoffman, G. Fernando, V. Goyal, and M. Civanlar, "RTP payload [14] 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.
[19] N. S. Jayant and P. Noll, Digital Coding of Waveforms-- [15] 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.
[20] K. McKay, "RTP Payload Format for PureVoice(tm) Audio", Request [16] K. McKay, "RTP Payload Format for PureVoice(tm) Audio", Request
for Comments (Proposed Standard) RFC 2658, Internet Engineering Task for Comments (Proposed Standard) RFC 2658, Internet Engineering Task
Force, Aug. 1999. Force, Aug. 1999.
[21] C. Perkins, I. Kouvelas, O. Hodson, V. Hardman, M. Handley, J.C. [17] 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.
[22] D. Tynan, "RTP payload format for BT.656 Video Encoding," [18] M. Speer and D. Hoffman, "RTP payload format of sun's CellB
Request for Comments (Proposed Standard) RFC 2431, Internet
Engineering Task Force, Oct. 1998.
[23] M. Speer and D. Hoffman, "RTP payload format of sun's CellB
video encoding," Request for Comments (Proposed Standard) RFC 2029, video encoding," Request for Comments (Proposed Standard) RFC 2029,
Internet Engineering Task Force, Oct. 1996. Internet Engineering Task Force, Oct. 1996.
[24] L. Berc, W. Fenner, R. Frederick, and S. McCanne, "RTP payload [19] L. Berc, W. Fenner, R. Frederick, and S. McCanne, "RTP payload
format for JPEG-compressed video," Request for Comments (Proposed format for JPEG-compressed video," Request for Comments (Proposed
Standard) RFC 2435, Internet Engineering Task Force, Oct. 1996. Standard) RFC 2435, Internet Engineering Task Force, Oct. 1996.
[25] T. Turletti and C. Huitema, "RTP payload format for H.261 video [20] T. Turletti and C. Huitema, "RTP payload format for H.261 video
streams," Request for Comments (Proposed Standard) RFC 2032, Internet streams," Request for Comments (Proposed Standard) RFC 2032, Internet
Engineering Task Force, Oct. 1996. Engineering Task Force, Oct. 1996.
[26] C. Zhu, "RTP payload format for H.263 video streams," Request [21] C. Bormann, L. Cline, G. Deisher, T. Gardos, C. Maciocco, D.
for Comments (Proposed Standard) RFC 2190, Internet Engineering Task
Force, Sep. 1997.
[27] C. Bormann, L. Cline, G. Deisher, T. Gardos, C. Maciocco, D.
Newell, J. Ott, G. Sullivan, S. Wenger, C. Zhu, "RTP Payload Format Newell, J. Ott, G. Sullivan, S. Wenger, C. Zhu, "RTP Payload Format
for the 1998 Version of ITU-T Rec. H.263 Video (H.263+)," Request for for the 1998 Version of ITU-T Rec. H.263 Video (H.263+)," Request for
Comments (Proposed Standard) RFC 2429, Internet Engineering Task Comments (Proposed Standard) RFC 2429, Internet Engineering Task
Force, Oct. 1998. Force, Oct. 1998.
[28] M. Civanlar, G. Cash, B. Haskell, "RTP Payload Format for [22] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming
Bundled MPEG," Request for Comments (Experimental) RFC 2343, Internet
Engineering Task Force, May 1998.
[29] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming
protocol (RTSP)," Request for Comments (Proposed Standard) RFC 2326, protocol (RTSP)," Request for Comments (Proposed Standard) RFC 2326,
Internet Engineering Task Force, Apr. 1998. Internet Engineering Task Force, Apr. 1998.
[30] S. Deering, "Host Extensions for IP Multicasting," Request for [23] S. Deering, "Host Extensions for IP Multicasting," Request for
Comments RFC 1112, STD 5, Internet Engineering Task Force, Aug. 1989. Comments RFC 1112, STD 5, Internet Engineering Task Force, Aug. 1989.
Current Locations of Related Resources Current Locations of Related Resources
Note: Several sections below refer to the ITU-T Software Tool Library Note: Several sections below refer to the ITU-T Software Tool Library
(STL). It is available from the ITU Sales Service, Place des Nations, (STL). It is available from the ITU Sales Service, Place des Nations,
CH-1211 Geneve 20, Switzerland (also check http://www.itu.int. The CH-1211 Geneve 20, Switzerland (also check http://www.itu.int. The
ITU-T STL is covered by a license defined in ITU-T Recommendation ITU-T STL is covered by a license defined in ITU-T Recommendation
G.191, "Software tools for speech and audio coding standardization". G.191, "Software tools for speech and audio coding standardization".
UTF-8 UTF-8
Information on the UCS Transformation Format 8 (UTF-8) is available Information on the UCS Transformation Format 8 (UTF-8) is available
at at
http://www.stonehand.com/unicode/standard/utf8.html http://www.stonehand.com/unicode/standard/utf8.html
1016
The U.S. DoD's Federal-Standard-1016 based 4800 bps code excited
linear prediction voice coder version 3.2 (CELP 3.2) Fortran and C
simulation source codes are available for worldwide distribution at
no charge (on DOS diskettes, but configured to compile on Sun SPARC
stations) from: Bob Fenichel, National Communications System,
Washington, D.C. 20305, phone +1-703-692-2124, fax +1-703-746-4960.
An implementation is also available at
ftp://ftp.super.org/pub/speech/celp_3.2a.tar.Z
DVI4 DVI4
An implementation is available from Jack Jansen at An implementation is available from Jack Jansen at
ftp://ftp.cwi.nl/local/pub/audio/adpcm.shar ftp://ftp.cwi.nl/local/pub/audio/adpcm.shar
G722 G722
An implementation of the G.722 algorithm is available as part of the An implementation of the G.722 algorithm is available as part of the
ITU-T STL, described above. ITU-T STL, described above.
G723 G726
The reference C code implementation defining the G.723.1 algorithm
and its Annexes A, B, and C are available as an integral part of
Recommendation G.723.1 from the ITU Sales Service, address listed
above. Both the algorithm and C code are covered by a specific
license. The ITU-T Secretariat should be contacted to obtain such
licensing information.
G726-32
G726-32 is specified in the ITU-T Recommendation G.726, "40, 32, 24, G726 is specified in the ITU-T Recommendation G.726, "40, 32, 24,
and 16 kb/s Adaptive Differential Pulse Code Modulation (ADPCM)". An and 16 kb/s Adaptive Differential Pulse Code Modulation (ADPCM)". An
implementation of the G.726 algorithm is available as part of the implementation of the G.726 algorithm is available as part of the
ITU-T STL, described above. ITU-T STL, described above.
G729 G729
The reference C code implementation defining the G.729 algorithm and The reference C code implementation defining the G.729 algorithm and
its Annexes A through I are available as an integral part of its Annexes A through I are available as an integral part of
Recommendation G.729 from the ITU Sales Service, listed above. Annex Recommendation G.729 from the ITU Sales Service, listed above. Annex
I contains the integrated C source code for all G.729 operating I contains the integrated C source code for all G.729 operating
skipping to change at page 43, line 27 skipping to change at page 38, line 27
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 [13]. companded data is also available in [9].
Table of Contents Table of Contents
1 Introduction ........................................ 3 1 Introduction ........................................ 3
1.1 Terminology ......................................... 3 1.1 Terminology ......................................... 4
2 RTP and RTCP Packet Forms and Protocol Behavior ..... 4 2 RTP and RTCP Packet Forms and Protocol Behavior ..... 4
3 IANA Considerations ................................. 7 3 IANA Considerations ................................. 7
3.1 Registering Additional Encodings .................... 7 3.1 Registering Additional Encodings .................... 7
4 Audio ............................................... 8 4 Audio ............................................... 8
4.1 Encoding-Independent Rules .......................... 8 4.1 Encoding-Independent Rules .......................... 8
4.2 Operating Recommendations ........................... 10 4.2 Operating Recommendations ........................... 10
4.3 Guidelines for Sample-Based Audio Encodings ......... 10 4.3 Guidelines for Sample-Based Audio Encodings ......... 10
4.4 Guidelines for Frame-Based Audio Encodings .......... 11 4.4 Guidelines for Frame-Based Audio Encodings .......... 11
4.5 Audio Encodings ..................................... 11 4.5 Audio Encodings ..................................... 11
4.5.1 1016 ................................................ 12 4.5.1 DVI4 ................................................ 12
4.5.2 DVI4 ................................................ 12 4.5.2 G722 ................................................ 14
4.5.3 G722 ................................................ 14 4.5.3 G726-40, G726-32, G726-24, and G726-16............... 14
4.5.4 G723 ................................................ 14 4.5.4 G728 ................................................ 16
4.5.5 G726-32 ............................................. 18 4.5.5 G729 ................................................ 16
4.5.6 G728 ................................................ 18 4.5.6 G729D and G729E ..................................... 18
4.5.7 G729 ................................................ 19 4.5.7 GSM ................................................. 21
4.5.8 G729D and G729E ..................................... 21 4.5.7.1 General Packaging Issues ............................ 21
4.5.9 GSM ................................................. 24 4.5.7.2 GSM variable names and numbers ...................... 21
4.5.9.1 General Packaging Issues ............................ 24 4.5.8 L8 .................................................. 21
4.5.9.2 GSM variable names and numbers ...................... 25 4.5.9 L16 ................................................. 22
4.5.10 GSM-HR .............................................. 25 4.5.10 LPC ................................................. 23
4.5.11 GSM-EFR ............................................. 25 4.5.11 MPA ................................................. 24
4.5.12 L8 .................................................. 25 4.5.12 PCMA and PCMU ....................................... 24
4.5.13 L16 ................................................. 26 4.5.13 QCELP ............................................... 24
4.5.14 LPC ................................................. 27 4.5.14 RED ................................................. 24
4.5.15 MPA ................................................. 28 4.5.15 VDVI ................................................ 25
4.5.16 PCMA and PCMU ....................................... 28 5 Video ............................................... 25
4.5.17 QCELP ............................................... 28 5.1 CelB ................................................ 26
4.5.18 RED ................................................. 28 5.2 JPEG ................................................ 26
4.5.19 VDVI ................................................ 29 5.3 H261 ................................................ 26
5 Video ............................................... 29 5.4 H263-1998 ........................................... 27
5.1 BT656 ............................................... 30 5.5 MPV ................................................. 27
5.2 CelB ................................................ 30 5.8 nv .................................................. 27
5.3 JPEG ................................................ 30 6 Payload Type Definitions ............................ 28
5.4 H261 ................................................ 30 7 RTP over TCP and Similar Byte Stream Protocols ...... 30
5.5 H263 ................................................ 31 8 Port Assignment ..................................... 30
5.6 H263-1998 ........................................... 31 9 Changes from RFC 1890 ............................... 31
5.7 MPV ................................................. 31 10 Security Considerations ............................. 33
5.8 MP2T ................................................ 31 11 Full Copyright Statement ............................ 33
5.9 MP1S ................................................ 31 12 Acknowledgements .................................... 34
5.10 MP2P ................................................ 31 13 Addresses of Authors ................................ 34
5.11 BMPEG ............................................... 31 A Bibliography ........................................ 34
5.12 nv .................................................. 32
6 Payload Type Definitions ............................ 32
7 RTP over TCP and Similar Byte Stream Protocols ...... 34
8 Port Assignment ..................................... 34
9 Changes from RFC 1890 ............................... 35
10 Security Considerations ............................. 37
11 Full Copyright Statement ............................ 37
12 Acknowledgements .................................... 38
13 Addresses of Authors ................................ 38
A Bibliography ........................................ 38
 End of changes. 

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