Network Working Group A. Sollaud
Request for Comments: 4749 France Telecom
Category: Standards Track October 2006 RTP Payload Format for the G.729.1 Audio Codec
Status of this Memo
This document specifies an Internet standards track protocol for the
Internet community, and requests discussion and suggestions for
improvements. Please refer to the current edition of the "Internet
Official Protocol Standards" (STD 1) for the standardization state
and status of this protocol. Distribution of this memo is unlimited.
Copyright (C) The Internet Society (2006).
This document specifies a Real-time Transport Protocol (RTP) payload
format to be used for the International Telecommunication Union
(ITU-T) G.729.1 audio codec. A media type registration is included
for this payload format.
Table of Contents
1. Introduction ....................................................22. Background ......................................................23. Embedded Bit Rates Considerations ...............................34. RTP Header Usage ................................................35. Payload Format ..................................................45.1. Payload Structure ..........................................45.2. Payload Header: MBS Field ..................................45.3. Payload Header: FT Field ...................................65.4. Audio Data .................................................66. Payload Format Parameters .......................................76.1. Media Type Registration ....................................76.2. Mapping to SDP Parameters ..................................86.2.1. Offer-Answer Model Considerations ...................96.2.2. Declarative SDP Considerations .....................117. Congestion Control .............................................118. Security Considerations ........................................119. IANA Considerations ............................................1210. References ....................................................1210.1. Normative References .....................................1210.2. Informative References ...................................12
The International Telecommunication Union (ITU-T) recommendation
G.729.1  is a scalable and wideband extension of the
recommendation G.729  audio codec. This document specifies the
payload format for packetization of G.729.1 encoded audio signals
into the Real-time Transport Protocol (RTP).
The payload format itself is described in Section 5. A media type
registration and the details for the use of G.729.1 with SDP are
given in Section 6.
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT","RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 .
G.729.1 is an 8-32 kbps scalable wideband (50-7000 Hz) speech and
audio coding algorithm interoperable with G.729, G.729 Annex A, and
G.729 Annex B. It provides a standardized solution for packetized
voice applications that allows a smooth transition from narrowband to
The most important services addressed are IP telephony and
videoconferencing, either for enterprise corporate networks or for
mass market (like Public Switched Telephone Network (PSTN) emulation
over DSL or wireless access). Target devices can be IP phones or
other VoIP handsets, home gateways, media gateways, IP Private Branch
Exchange (IPBX), trunking equipment, voice messaging servers, etc.
For all those applications, the scalability feature allows tuning the
bit rate versus quality trade-off, possibly in a dynamic way during a
session, taking into account service requirements and network
The G.729.1 coder produces an embedded bitstream structured in 12
layers corresponding to 12 available bit rates between 8 and 32 kbps.
The first layer, at 8 kbps, is called the core layer and is bitstream
compatible with the ITU-T G.729/G.729A coder. At 12 kbps, a second
layer improves the narrowband quality. Upper layers provide wideband
audio (50-7000 Hz) between 14 and 32 kbps, with a 2 kbps granularity
allowing graceful quality improvements. Only the core layer is
mandatory to decode understandable speech; upper layers provide
quality enhancement and wideband enlargement.
The codec operates on 20-ms frames, and the default sampling rate is
16 kHz. Input and output at 8 kHz are also supported, at all bit
3. Embedded Bit Rates Considerations
The embedded property of G.729.1 streams provides a mechanism to
adjust the bandwidth demand. At any time, a sender can change its
sending bit rate without external signalling, and the receiver will
be able to properly decode the frames. It may help to control
congestion, since the bandwidth can be adjusted by selecting another
The ability to adjust the bandwidth may also help when having a fixed
bandwidth link dedicated to voice calls, for example in a residential
or trunking gateway. In that case, the system can change the bit
rates depending on the number of simultaneous calls. This will only
impact the sending bandwidth. In order to adjust the receiving
bandwidth as well, we introduce an in-band signalling to request the
other party to change its own sending bit rate. This in-band request
is called MBS, for Maximum Bit rate Supported. It is described in
Section 5.2. Note that it is only useful for two-way unicast G.729.1
traffic, because when A sends an in-band MBS to B in order to request
that B modify its sending bit rate, it concerns the stream from B to
A. If there is no G.729.1 stream in the reverse direction, the MBS
will have no effect.
4. RTP Header Usage
The format of the RTP header is specified in RFC 3550 . This
payload format uses the fields of the header in a manner consistent
with that specification.
The RTP timestamp clock frequency is the same as the default sampling
frequency: 16 kHz.
G.729.1 has also the capability to operate with 8 kHz sampled input/
output signals at all bit rates. It does not affect the bitstream,
and the decoder does not require a priori knowledge about the
sampling rate of the original signal at the input of the encoder.
Therefore, depending on the implementation and the audio acoustic
capabilities of the devices, the input of the encoder and/or the
output of the decoder can be configured at 8 kHz; however, a 16 kHz
RTP clock rate MUST always be used.
The duration of one frame is 20 ms, corresponding to 320 samples at
16 kHz. Thus the timestamp is increased by 320 for each consecutive
The M bit MUST be set to zero in all packets.
The assignment of an RTP payload type for this packet format is
outside the scope of the document, and will not be specified here.
It is expected that the RTP profile under which this payload format
is being used will assign a payload type for this codec or specify
that the payload type is to be bound dynamically (see Section 6.2).
5. Payload Format
5.1. Payload Structure
The complete payload consists of a payload header of 1 octet,
followed by zero or more consecutive audio frames at the same bit
The payload header consists of two fields: MBS (see Section 5.2) and
FT (see Section 5.3).
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
| MBS | FT | |
: zero or more frames at the same bit rate :
5.2. Payload Header: MBS Field
MBS (4 bits): maximum bit rate supported. Indicates a maximum bit
rate to the encoder at the site of the receiver of this payload. The
value of the MBS field is set according to the following table:
| MBS | max bit rate |
| 0 | 8 kbps |
| 1 | 12 kbps |
| 2 | 14 kbps |
| 3 | 16 kbps |
| 4 | 18 kbps |
| 5 | 20 kbps |
| 6 | 22 kbps |
| 7 | 24 kbps |
| 8 | 26 kbps |
| 9 | 28 kbps |
| 10 | 30 kbps |
| 11 | 32 kbps |
| 12-14 | (reserved) |
| 15 | NO_MBS |
The MBS is used to tell the other party the maximum bit rate one can
receive. The encoder MUST NOT exceed the sending rate indicated by
the received MBS. Note that, due to the embedded property of the
coding scheme, the encoder can send frames at the MBS rate or any
lower rate. As long as it does not exceed the MBS, the encoder can
change its bit rate at any time without previous notice.
Note that the MBS is a codec bit rate; the actual network bit rate is
higher and depends on the overhead of the underlying protocols.
The MBS received is valid until the next MBS is received, i.e., a
newly received MBS value overrides the previous one.
If a payload with a reserved MBS value is received, the MBS MUST be
The MBS field MUST be set to 15 for packets sent to a multicast group
and MUST be ignored on packets received from a multicast group.
The MBS field MUST be set to 15 in all packets when the actual MBS
value is sent through non-RTP means. This is out of the scope of
See Sections 3 and 7 for more details on the use of MBS for
5.3. Payload Header: FT Field
FT (4 bits): Frame type of the frame(s) in this packet, as per the
| FT | encoding rate | frame size |
| 0 | 8 kbps | 20 octets |
| 1 | 12 kbps | 30 octets |
| 2 | 14 kbps | 35 octets |
| 3 | 16 kbps | 40 octets |
| 4 | 18 kbps | 45 octets |
| 5 | 20 kbps | 50 octets |
| 6 | 22 kbps | 55 octets |
| 7 | 24 kbps | 60 octets |
| 8 | 26 kbps | 65 octets |
| 9 | 28 kbps | 70 octets |
| 10 | 30 kbps | 75 octets |
| 11 | 32 kbps | 80 octets |
| 12-14 | (reserved) | |
| 15 | NO_DATA | 0 |
The FT value 15 (NO_DATA) indicates that there is no audio data in
the payload. This MAY be used to update the MBS value when there is
no audio frame to transmit. The payload will then be reduced to the
If a payload with a reserved FT value is received, the whole payload
MUST be ignored.
5.4. Audio Data
Audio data of a payload contains one or more consecutive audio frames
at the same bit rate. The audio frames are packed in order of time,
that is, oldest first.
The size of one frame is given by the FT field, as per the table in
Section 5.3, and the actual number of frames is easy to infer from
the size of the audio data part:
nb_frames = (size_of_audio_data) / (size_of_one_frame).
Only full frames must be considered. So if there is a remainder to
the division above, the corresponding remaining bytes in the received
payload MUST be ignored.
Note that if FT=15, there will be no audio frame in the payload.
6. Payload Format Parameters
This section defines the parameters that may be used to configure
optional features in the G.729.1 RTP transmission.
The parameters are defined here as part of the media subtype
registration for the G.729.1 codec. A mapping of the parameters into
the Session Description Protocol (SDP)  is also provided for those
applications that use SDP. In control protocols that do not use MIME
or SDP, the media type parameters must be mapped to the appropriate
format used with that control protocol.
6.1. Media Type Registration
This registration is done using the template defined in RFC 4288 
and following RFC 3555 .
Type name: audio
Subtype name: G7291
Required parameters: none
maxbitrate: the absolute maximum codec bit rate for the session, in
bits per second. Permissible values are 8000, 12000, 14000,
16000, 18000, 20000, 22000, 24000, 26000, 28000, 30000, and 32000.
32000 is implied if this parameter is omitted. The maxbitrate
restricts the range of bit rates which can be used. The bit rates
indicated by FT and MBS fields in the RTP packets MUST NOT exceed
mbs: the current maximum codec bit rate supported as a receiver, in
bits per second. Permissible values are in the same set as for
the maxbitrate parameter, with the constraint that mbs MUST be
lower or equal to maxbitrate. If the mbs parameter is omitted, it
is set to the maxbitrate value. So if both mbs and maxbitrate are
omitted, they are both set to 32000. The mbs parameter
corresponds to a MBS value in the RTP packets as per table in
Section 5.2 of RFC 4749. Note that this parameter may be
dynamically updated by the MBS field of the RTP packets sent; it
is not an absolute value for the session.
ptime: the recommended length of time (in milliseconds) represented
by the media in a packet. See Section 6 of RFC 4566 .
maxptime: the maximum length of time (in milliseconds) that can be
encapsulated in a packet. See Section 6 of RFC 4566 
Encoding considerations: This media type is framed and contains
binary data; see Section 4.8 of RFC 4288 .
Security considerations: See Section 8 of RFC 4749
Interoperability considerations: none
Published specification: RFC 4749
Applications which use this media type: Audio and video conferencing
Additional information: none
Person & email address to contact for further information:
Aurelien Sollaud, firstname.lastname@example.org
Intended usage: COMMON
Restrictions on usage: This media type depends on RTP framing, and
hence is only defined for transfer via RTP .
Author: Aurelien Sollaud
Change controller: IETF Audio/Video Transport working group delegated
from the IESG
6.2. Mapping to SDP Parameters
The information carried in the media type specification has a
specific mapping to fields in the Session Description Protocol (SDP)
, which is commonly used to describe RTP sessions. When SDP is
used to specify sessions employing the G.729.1 codec, the mapping is
o The media type ("audio") goes in SDP "m=" as the media name.
o The media subtype ("G7291") goes in SDP "a=rtpmap" as the encoding
name. The RTP clock rate in "a=rtpmap" MUST be 16000 for G.729.1.
o The parameters "ptime" and "maxptime" go in the SDP "a=ptime" and
"a=maxptime" attributes, respectively.
o Any remaining parameters go in the SDP "a=fmtp" attribute by
copying them directly from the media type string as a semicolon
separated list of parameter=value pairs.
Some example SDP session descriptions utilizing G.729.1 encodings
Example 1: default parameters
m=audio 53146 RTP/AVP 98
Example 2: recommended packet duration of 40 ms (=2 frames), maximum
bit rate is 12 kbps, and initial MBS set to 8 kbps. It could be a
loaded PSTN gateway which can operate at 12 kbps but asks to
initially reduce the bit rate to 8 kbps.
m=audio 51258 RTP/AVP 99
a=fmtp:99 maxbitrate=12000; mbs=8000
6.2.1. Offer-Answer Model Considerations
The following considerations apply when using SDP offer-answer
procedures  to negotiate the use of G.729.1 payload in RTP:
o Since G.729.1 is an extension of G.729, the offerer SHOULD
announce G.729 support in its "m=audio" line, with G.729.1
preferred. This will allow interoperability with both G.729.1 and
G.729-only capable parties.
Below is an example of such an offer:
m=audio 55954 RTP/AVP 98 18
If the answerer supports G.729.1, it will keep the payload type 98
in its answer, and the conversation will be done using G.729.1.
Else, if the answerer supports only G.729, it will leave only the
payload type 18 in its answer, and the conversation will be done
using G.729 (the payload format for G.729 is defined in Section
4.5.6 of RFC 3551 ).
Note that when used at 8 kbps in G.729-compatible mode, the
G.729.1 decoder supports G.729 Annex B. Therefore, Annex B can be
advertised (by default, annexb=yes for G729 media type; see
Section 4.1.9 of RFC 3555 ).
o The "maxbitrate" parameter is bi-directional. If the offerer sets
a maxbitrate value, the answerer MUST reply with a smaller or
equal value. The actual maximum bit rate for the session will be
o If the received value for "maxbitrate" is between 8000 and 32000
but not in the permissible values set, it SHOULD be read as the
closest lower valid value. If the received value is lower than
8000 or greater than 32000, the session MUST be rejected.
o The "mbs" parameter is not symmetric. Values in the offer and the
answer are independent and take into account local constraints.
One party MUST NOT start sending frames at a bit rate higher than
the "mbs" of the other party. The parameter allows announcing
this value, prior to the sending of any packet, to prevent the
remote sender from exceeding the MBS at the beginning of the
o If the received value for "mbs" is greater or equal to 8000 but
not in the permissible values set, it SHOULD be read as the
closest lower valid value. If the received value is lower than
8000, the session MUST be rejected.
o The parameters "ptime" and "maxptime" will in most cases not
affect interoperability. The SDP offer-answer handling of the
"ptime" parameter is described in RFC 3264 . The "maxptime"
parameter MUST be handled in the same way.
o Any unknown parameter in an offer MUST be ignored by the receiver
and MUST NOT be included in the answer.
Some special rules apply for mono-directional traffic:
o For sendonly streams, the "mbs" parameter is useless and SHOULD
NOT be used.
o For recvonly streams, the "mbs" parameter is the only way to
communicate the MBS to the sender, since there is no RTP stream
towards it. So to request a bit rate change, the receiver will
need to use an out-of-band mechanism, like a SIP RE-INVITE.
Some special rules apply for multicast:
o The "mbs" parameter MUST NOT be used.
o The "maxbitrate" parameter becomes declarative and MUST NOT be
negotiated. This parameter is fixed, and a participant MUST use
the configuration that is provided for the session.
6.2.2. Declarative SDP Considerations
For declarative use of SDP such as in SAP  and RTSP , the
following considerations apply:
o The "mbs" parameter MUST NOT be used.
o The "maxbitrate" parameter is declarative and provides the
parameter that SHALL be used when receiving and/or sending the
7. Congestion Control
Congestion control for RTP SHALL be used in accordance with RFC 3550
 and any appropriate profile (for example, RFC 3551 ). The
embedded and variable bit rates capability of G.729.1 provides a
mechanism that may help to control congestion; see Section 3 for more
The number of frames encapsulated in each RTP payload influences the
overall bandwidth of the RTP stream, due to the header overhead.
Packing more frames in each RTP payload can reduce the number of
packets sent and hence the header overhead, at the expense of
increased delay and reduced error robustness.
8. Security Considerations
RTP packets using the payload format defined in this specification
are subject to the general security considerations discussed in the
RTP specification  and any appropriate profile (for example, RFC
As this format transports encoded speech/audio, the main security
issues include confidentiality, integrity protection, and
authentication of the speech/audio itself. The payload format itself
does not have any built-in security mechanisms. Any suitable
external mechanisms, such as SRTP , MAY be used.
This payload format and the G.729.1 encoding do not exhibit any
significant non-uniformity in the receiver-end computational load and
thus are unlikely to pose a denial-of-service threat due to the
receipt of pathological datagrams.
9. IANA Considerations
IANA has registered audio/G7291 as a media subtype; see Section 6.1.
10.1. Normative References
 International Telecommunications Union, "G.729 based Embedded
Variable bit-rate coder: An 8-32 kbit/s scalable wideband coder
bitstream interoperable with G.729", ITU-T Recommendation
G.729.1, May 2006.
 Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
 Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
"RTP: A Transport Protocol for Real-Time Applications", STD 64,
RFC 3550, July 2003.
 Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video
Conferences with Minimal Control", STD 65, RFC 3551, July 2003.
 Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
 Freed, N. and J. Klensin, "Media Type Specifications and
Registration Procedures", BCP 13, RFC 4288, December 2005.
 Casner, S. and P. Hoschka, "MIME Type Registration of RTP
Payload Formats", RFC 3555, July 2003.
 Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
Session Description Protocol (SDP)", RFC 3264, June 2002.
10.2. Informative References
 International Telecommunications Union, "Coding of speech at 8
kbit/s using conjugate-structure algebraic-code-excited linear-
prediction (CS-ACELP)", ITU-T Recommendation G.729, March 1996.
 Handley, M., Perkins, C., and E. Whelan, "Session Announcement
Protocol", RFC 2974, October 2000.
 Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time Streaming
Protocol (RTSP)", RFC 2326, April 1998.
 Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
2 avenue Pierre Marzin
Lannion Cedex 22307
Phone: +33 2 96 05 15 06
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