draft-ietf-avt-rtp-g729-scal-wb-ext-07.txt   rfc4749.txt 
Network Working Group A. Sollaud Network Working Group A. Sollaud
Internet-Draft France Telecom Request for Comments: 4749 France Telecom
Intended status: Standards Track August 28, 2006 Category: Standards Track October 2006
Expires: March 1, 2007
RTP payload format for the G.729.1 audio codec RTP Payload Format for the G.729.1 Audio Codec
draft-ietf-avt-rtp-g729-scal-wb-ext-07
Status of this Memo Status of this Memo
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Copyright Notice Copyright Notice
Copyright (C) The Internet Society (2006). Copyright (C) The Internet Society (2006).
Abstract Abstract
This document specifies a real-time transport protocol (RTP) payload This document specifies a Real-time Transport Protocol (RTP) payload
format to be used for the International Telecommunication Union format to be used for the International Telecommunication Union
(ITU-T) G.729.1 audio codec. A media type registration is included (ITU-T) G.729.1 audio codec. A media type registration is included
for this payload format. for this payload format.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction ....................................................2
2. Background . . . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Background ......................................................2
3. Embedded bit rates considerations . . . . . . . . . . . . . . 4 3. Embedded Bit Rates Considerations ...............................3
4. RTP header usage . . . . . . . . . . . . . . . . . . . . . . . 4 4. RTP Header Usage ................................................3
5. Payload format . . . . . . . . . . . . . . . . . . . . . . . . 5 5. Payload Format ..................................................4
5.1. Payload structure . . . . . . . . . . . . . . . . . . . . 5 5.1. Payload Structure ..........................................4
5.2. Payload Header: MBS field . . . . . . . . . . . . . . . . 5 5.2. Payload Header: MBS Field ..................................4
5.3. Payload Header: FT field . . . . . . . . . . . . . . . . . 7 5.3. Payload Header: FT Field ...................................6
5.4. Audio data . . . . . . . . . . . . . . . . . . . . . . . . 7 5.4. Audio Data .................................................6
6. Payload format parameters . . . . . . . . . . . . . . . . . . 8 6. Payload Format Parameters .......................................7
6.1. Media type registration . . . . . . . . . . . . . . . . . 8 6.1. Media Type Registration ....................................7
6.2. Mapping to SDP parameters . . . . . . . . . . . . . . . . 9 6.2. Mapping to SDP Parameters ..................................8
6.2.1. Offer-answer model considerations . . . . . . . . . . 10 6.2.1. Offer-Answer Model Considerations ...................9
6.2.2. Declarative SDP considerations . . . . . . . . . . . . 12 6.2.2. Declarative SDP Considerations .....................11
7. Congestion control . . . . . . . . . . . . . . . . . . . . . . 12 7. Congestion Control .............................................11
8. Security considerations . . . . . . . . . . . . . . . . . . . 12 8. Security Considerations ........................................11
9. IANA considerations . . . . . . . . . . . . . . . . . . . . . 13 9. IANA Considerations ............................................12
10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 13 10. References ....................................................12
10.1. Normative references . . . . . . . . . . . . . . . . . . . 13 10.1. Normative References .....................................12
10.2. Informative references . . . . . . . . . . . . . . . . . . 13 10.2. Informative References ...................................12
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 14
Intellectual Property and Copyright Statements . . . . . . . . . . 15
1. Introduction 1. Introduction
The International Telecommunication Union (ITU-T) recommendation The International Telecommunication Union (ITU-T) recommendation
G.729.1 [1] is a scalable and wideband extension of the G.729.1 [1] is a scalable and wideband extension of the
recommendation G.729 [9] audio codec. This document specifies the recommendation G.729 [9] audio codec. This document specifies the
payload format for packetization of G.729.1 encoded audio signals payload format for packetization of G.729.1 encoded audio signals
into the real-time transport protocol (RTP). into the Real-time Transport Protocol (RTP).
The payload format itself is described in Section 5. A media type 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 registration and the details for the use of G.729.1 with SDP are
given in Section 6. given in Section 6.
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT","RECOMMENDED", "MAY", and "OPTIONAL" in this "SHOULD", "SHOULD NOT","RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [2]. document are to be interpreted as described in RFC 2119 [2].
2. Background 2. Background
G.729.1 is a 8-32 kbps scalable wideband (50-7000 Hz) speech and 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 audio coding algorithm interoperable with G.729, G.729 Annex A, and
G.729 Annex B. It provides a standardized solution for packetized G.729 Annex B. It provides a standardized solution for packetized
voice applications that allows a smooth transition from narrowband to voice applications that allows a smooth transition from narrowband to
wideband telephony. wideband telephony.
The most important services addressed are IP telephony and The most important services addressed are IP telephony and
videoconferencing, either for enterprise corporate networks or for videoconferencing, either for enterprise corporate networks or for
mass market (like PSTN emulation over DSL or wireless access). mass market (like Public Switched Telephone Network (PSTN) emulation
Target devices can be IP phones or other VoIP handsets, home over DSL or wireless access). Target devices can be IP phones or
gateways, media gateways, IPBX, trunking equipment, voice messaging other VoIP handsets, home gateways, media gateways, IP Private Branch
servers, etc. Exchange (IPBX), trunking equipment, voice messaging servers, etc.
For all those applications, the scalability feature allows to tune For all those applications, the scalability feature allows tuning the
the bit rate versus quality trade-off, possibly in a dynamic way bit rate versus quality trade-off, possibly in a dynamic way during a
during a session, taking into account service requirements and session, taking into account service requirements and network
network transport constraints. transport constraints.
The G.729.1 coder produces an embedded bitstream structured in 12 The G.729.1 coder produces an embedded bitstream structured in 12
layers corresponding to 12 available bit rates between 8 and 32 kbps. 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 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 compatible with the ITU-T G.729/G.729A coder. At 12 kbps, a second
layer improves the narrowband quality. Upper layers provides layer improves the narrowband quality. Upper layers provide wideband
wideband audio (50-7000 Hz) between 14 and 32 kbps, with a 2 kbps audio (50-7000 Hz) between 14 and 32 kbps, with a 2 kbps granularity
granularity allowing graceful quality improvements. Only the core allowing graceful quality improvements. Only the core layer is
layer is mandatory to decode understandable speech, upper layers mandatory to decode understandable speech; upper layers provide
provide quality enhancement and wideband enlargement. quality enhancement and wideband enlargement.
The codec operates on 20 ms frames, and the default sampling rate is 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 16 kHz. Input and output at 8 kHz are also supported, at all bit
rates. rates.
3. Embedded bit rates considerations 3. Embedded Bit Rates Considerations
The embedded property of G.729.1 streams provides a mechanism to The embedded property of G.729.1 streams provides a mechanism to
adjust the bandwidth demand. At any time, a sender can change its adjust the bandwidth demand. At any time, a sender can change its
sending bit rate without an external signalling, and the receiver sending bit rate without external signalling, and the receiver will
will be able to properly decode the frames. It may help to control be able to properly decode the frames. It may help to control
congestion, since the bandwidth can be adjusted by selecting another congestion, since the bandwidth can be adjusted by selecting another
bit rate. bit rate.
The ability to adjust the bandwidth may also help when having a fixed The ability to adjust the bandwidth may also help when having a fixed
bandwidth link dedicated to voice calls, for example in a residential bandwidth link dedicated to voice calls, for example in a residential
or trunking gateway. In that case, the system can change the bit or trunking gateway. In that case, the system can change the bit
rates depending on the number of simultaneous calls. Since it only rates depending on the number of simultaneous calls. This will only
impacts the sending bandwidth, we introduce an in-band signalling to impact the sending bandwidth. In order to adjust the receiving
request the other party to change its own sending bit rate, in order bandwidth as well, we introduce an in-band signalling to request the
to adjust the receiving bandwidth as well. This in-band request is other party to change its own sending bit rate. This in-band request
called MBS, for Maximum Bit rate Supported. It is described in 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 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, it is to request B traffic, because when A sends an in-band MBS to B in order to request
to modify its sending bit rate, that is for the stream from B to A. that B modify its sending bit rate, it concerns the stream from B to
If there is no G.729.1 stream in the reverse direction, the MBS will A. If there is no G.729.1 stream in the reverse direction, the MBS
have no effect. will have no effect.
4. RTP header usage 4. RTP Header Usage
The format of the RTP header is specified in RFC 3550 [3]. This The format of the RTP header is specified in RFC 3550 [3]. This
payload format uses the fields of the header in a manner consistent payload format uses the fields of the header in a manner consistent
with that specification. with that specification.
The RTP timestamp clock frequency is the same as the default sampling The RTP timestamp clock frequency is the same as the default sampling
frequency, that is 16 kHz. frequency: 16 kHz.
G.729.1 has also the capability to operate with 8 kHz sampled input/ 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 output signals at all bit rates. It does not affect the bitstream,
and the decoder does not require a priori knowledge about the and the decoder does not require a priori knowledge about the
sampling rate of the original signal at the input of the encoder. sampling rate of the original signal at the input of the encoder.
Therefore, depending on the implementation and the audio acoustic Therefore, depending on the implementation and the audio acoustic
capabilities of the devices, the input of the encoder and/or the 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 output of the decoder can be configured at 8 kHz; however, a 16 kHz
RTP clock rate MUST always be used. RTP clock rate MUST always be used.
The duration of one frame is 20 ms, corresponding to 320 samples at 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 16 kHz. Thus the timestamp is increased by 320 for each consecutive
frame. frame.
The M bit MUST be set to zero in all packets. The M bit MUST be set to zero in all packets.
The assignment of an RTP payload type for this packet format is The assignment of an RTP payload type for this packet format is
outside the scope of the document, and will not be specified here. outside the scope of the document, and will not be specified here.
It is expected that the RTP profile under which this payload format 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 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). that the payload type is to be bound dynamically (see Section 6.2).
5. Payload format 5. Payload Format
5.1. Payload structure 5.1. Payload Structure
The complete payload consists of a payload header of 1 octet, The complete payload consists of a payload header of 1 octet,
followed by zero or more consecutive audio frames at the same bit followed by zero or more consecutive audio frames at the same bit
rate. rate.
The payload header consists of two fields: MBS (see Section 5.2) and The payload header consists of two fields: MBS (see Section 5.2) and
FT (see Section 5.3). FT (see Section 5.3).
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| MBS | FT | | | MBS | FT | |
+-+-+-+-+-+-+-+-+ + +-+-+-+-+-+-+-+-+ +
: zero or more frames at the same bit rate : : zero or more frames at the same bit rate :
: : : :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
5.2. Payload Header: MBS field 5.2. Payload Header: MBS Field
MBS (4 bits): maximum bit rate supported. Indicates a maximum bit 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 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: value of the MBS field is set according to the following table:
+-------+--------------+ +-------+--------------+
| MBS | max bit rate | | MBS | max bit rate |
+-------+--------------+ +-------+--------------+
| 0 | 8 kbps | | 0 | 8 kbps |
| 1 | 12 kbps | | 1 | 12 kbps |
skipping to change at page 6, line 26 skipping to change at page 5, line 26
| 8 | 26 kbps | | 8 | 26 kbps |
| 9 | 28 kbps | | 9 | 28 kbps |
| 10 | 30 kbps | | 10 | 30 kbps |
| 11 | 32 kbps | | 11 | 32 kbps |
| 12-14 | (reserved) | | 12-14 | (reserved) |
| 15 | NO_MBS | | 15 | NO_MBS |
+-------+--------------+ +-------+--------------+
The MBS is used to tell the other party the maximum bit rate one can 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 receive. The encoder MUST NOT exceed the sending rate indicated by
the received MBS. Thanks to the embedded property of the coding the received MBS. Note that, due to the embedded property of the
scheme, note that it can send frames at the MBS rate or any lower coding scheme, the encoder can send frames at the MBS rate or any
rate. As long as it does not exceed the MBS, it can change its bit lower rate. As long as it does not exceed the MBS, the encoder can
rate at any time without previous notice. 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 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. higher and depends on the overhead of the underlying protocols.
The MBS received is valid until the next MBS is received, i.e. a The MBS received is valid until the next MBS is received, i.e., a
newly received MBS value overrides the previous one. newly received MBS value overrides the previous one.
If a payload with a reserved MBS value is received, the MBS MUST be If a payload with a reserved MBS value is received, the MBS MUST be
ignored. ignored.
The MBS field MUST be set to 15 for packets sent to a multicast The MBS field MUST be set to 15 for packets sent to a multicast group
group, and MUST be ignored on packets received from a multicast and MUST be ignored on packets received from a multicast group.
group.
The MBS field MUST be set to 15 in all packets when the actual MBS 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 value is sent through non-RTP means. This is out of the scope of
this specification. this specification.
See Section 3 and Section 7 for more details on the use of MBS for See Sections 3 and 7 for more details on the use of MBS for
congestion control. congestion control.
5.3. Payload Header: FT field 5.3. Payload Header: FT Field
FT (4 bits): Frame type of the frame(s) in this packet, as per the FT (4 bits): Frame type of the frame(s) in this packet, as per the
following table: following table:
+-------+---------------+------------+ +-------+---------------+------------+
| FT | encoding rate | frame size | | FT | encoding rate | frame size |
+-------+---------------+------------+ +-------+---------------+------------+
| 0 | 8 kbps | 20 octets | | 0 | 8 kbps | 20 octets |
| 1 | 12 kbps | 30 octets | | 1 | 12 kbps | 30 octets |
| 2 | 14 kbps | 35 octets | | 2 | 14 kbps | 35 octets |
skipping to change at page 7, line 37 skipping to change at page 6, line 37
+-------+---------------+------------+ +-------+---------------+------------+
The FT value 15 (NO_DATA) indicates that there is no audio data in 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 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 no audio frame to transmit. The payload will then be reduced to the
payload header. payload header.
If a payload with a reserved FT value is received, the whole payload If a payload with a reserved FT value is received, the whole payload
MUST be ignored. MUST be ignored.
5.4. Audio data 5.4. Audio Data
Audio data of a payload contains one or more consecutive audio frames 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, at the same bit rate. The audio frames are packed in order of time,
that is the older first. that is, oldest first.
The size of one frame is given by the FT field, as per the table in 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 Section 5.3, and the actual number of frames is easy to infer from
the size of the audio data part: the size of the audio data part:
nb_frames = (size_of_audio_data) / (size_of_one_frame). nb_frames = (size_of_audio_data) / (size_of_one_frame).
Only full frames must be considered. So if there is a remainder to Only full frames must be considered. So if there is a remainder to
the division above, the corresponding remaining bytes in the received the division above, the corresponding remaining bytes in the received
payload MUST be ignored. payload MUST be ignored.
Note that if FT=15, there will be no audio frame in the payload. Note that if FT=15, there will be no audio frame in the payload.
6. Payload format parameters 6. Payload Format Parameters
This section defines the parameters that may be used to configure This section defines the parameters that may be used to configure
optional features in the G.729.1 RTP transmission. optional features in the G.729.1 RTP transmission.
The parameters are defined here as part of the media subtype The parameters are defined here as part of the media subtype
registration for the G.729.1 codec. A mapping of the parameters into registration for the G.729.1 codec. A mapping of the parameters into
the Session Description Protocol (SDP) [5] is also provided for those the Session Description Protocol (SDP) [5] is also provided for those
applications that use SDP. In control protocols that do not use MIME applications that use SDP. In control protocols that do not use MIME
or SDP, the media type parameters must be mapped to the appropriate or SDP, the media type parameters must be mapped to the appropriate
format used with that control protocol. format used with that control protocol.
6.1. Media type registration 6.1. Media Type Registration
[Note to RFC Editor: Please replace all occurrences of RFC XXXX by
the RFC number assigned to this document]
This registration is done using the template defined in RFC 4288 [6] This registration is done using the template defined in RFC 4288 [6]
and following RFC 3555 [7]. and following RFC 3555 [7].
Type name: audio Type name: audio
Subtype name: G7291 Subtype name: G7291
Required parameters: none Required parameters: none
skipping to change at page 8, line 50 skipping to change at page 7, line 47
indicated by FT and MBS fields in the RTP packets MUST NOT exceed indicated by FT and MBS fields in the RTP packets MUST NOT exceed
maxbitrate. maxbitrate.
mbs: the current maximum codec bit rate supported as a receiver, in mbs: the current maximum codec bit rate supported as a receiver, in
bits per second. Permissible values are in the same set as for bits per second. Permissible values are in the same set as for
the maxbitrate parameter, with the constraint that mbs MUST be the maxbitrate parameter, with the constraint that mbs MUST be
lower or equal to maxbitrate. If the mbs parameter is omitted, it 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 is set to the maxbitrate value. So if both mbs and maxbitrate are
omitted, they are both set to 32000. The mbs parameter omitted, they are both set to 32000. The mbs parameter
corresponds to a MBS value in the RTP packets as per table in corresponds to a MBS value in the RTP packets as per table in
Section 5.2 of RFC XXXX. Note that this parameter may be Section 5.2 of RFC 4749. Note that this parameter may be
dynamically updated by the MBS field of the RTP packets sent, it dynamically updated by the MBS field of the RTP packets sent; it
is not an absolute value for the session. is not an absolute value for the session.
ptime: the recommended length of time in milliseconds represented by ptime: the recommended length of time (in milliseconds) represented
the media in a packet. See Section 6 of RFC 4566 [5]. by the media in a packet. See Section 6 of RFC 4566 [5].
maxptime: the maximum length of time in milliseconds which can be maxptime: the maximum length of time (in milliseconds) that can be
encapsulated in a packet. See Section 6 of RFC 4566 [5] encapsulated in a packet. See Section 6 of RFC 4566 [5]
Encoding considerations: This media type is framed and contains Encoding considerations: This media type is framed and contains
binary data, see section 4.8 of RFC 4288 [6]. binary data; see Section 4.8 of RFC 4288 [6].
Security considerations: See Section 8 of RFC XXXX Security considerations: See Section 8 of RFC 4749
Interoperability considerations: none Interoperability considerations: none
Published specification: RFC XXXX Published specification: RFC 4749
Applications which use this media type: Audio and video conferencing Applications which use this media type: Audio and video conferencing
tools. tools.
Additional information: none Additional information: none
Person & email address to contact for further information: Aurelien Person & email address to contact for further information:
Sollaud, aurelien.sollaud@orange-ft.com Aurelien Sollaud, aurelien.sollaud@orange-ftgroup.com
Intended usage: COMMON Intended usage: COMMON
Restrictions on usage: This media type depends on RTP framing, and Restrictions on usage: This media type depends on RTP framing, and
hence is only defined for transfer via RTP [3]. hence is only defined for transfer via RTP [3].
Author: Aurelien Sollaud Author: Aurelien Sollaud
Change controller: IETF Audio/Video Transport working group delegated Change controller: IETF Audio/Video Transport working group delegated
from the IESG from the IESG
6.2. Mapping to SDP parameters 6.2. Mapping to SDP Parameters
The information carried in the media type specification has a The information carried in the media type specification has a
specific mapping to fields in the Session Description Protocol (SDP) specific mapping to fields in the Session Description Protocol (SDP)
[5], which is commonly used to describe RTP sessions. When SDP is [5], which is commonly used to describe RTP sessions. When SDP is
used to specify sessions employing the G.729.1 codec, the mapping is used to specify sessions employing the G.729.1 codec, the mapping is
as follows: as follows:
o The media type ("audio") goes in SDP "m=" as the media name. 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 o The media subtype ("G7291") goes in SDP "a=rtpmap" as the encoding
skipping to change at page 10, line 23 skipping to change at page 9, line 20
follow. follow.
Example 1: default parameters Example 1: default parameters
m=audio 53146 RTP/AVP 98 m=audio 53146 RTP/AVP 98
a=rtpmap:98 G7291/16000 a=rtpmap:98 G7291/16000
Example 2: recommended packet duration of 40 ms (=2 frames), maximum 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 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 loaded PSTN gateway which can operate at 12 kbps but asks to
initially reduce the bit rate at 8 kbps. initially reduce the bit rate to 8 kbps.
m=audio 51258 RTP/AVP 99 m=audio 51258 RTP/AVP 99
a=rtpmap:99 G7291/16000 a=rtpmap:99 G7291/16000
a=fmtp:99 maxbitrate=12000; mbs=8000 a=fmtp:99 maxbitrate=12000; mbs=8000
a=ptime:40 a=ptime:40
6.2.1. Offer-answer model considerations 6.2.1. Offer-Answer Model Considerations
The following considerations apply when using SDP offer-answer The following considerations apply when using SDP offer-answer
procedures [8] to negotiate the use of G.729.1 payload in RTP: procedures [8] 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 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 announce G.729 support in its "m=audio" line, with G.729.1
preferred. This will allow interoperability with both G.729.1 and preferred. This will allow interoperability with both G.729.1 and
G.729-only capable parties. G.729-only capable parties.
Below is an example of such an offer: Below is an example of such an offer:
m=audio 55954 RTP/AVP 98 18 m=audio 55954 RTP/AVP 98 18
a=rtpmap:98 G7291/16000 a=rtpmap:98 G7291/16000
a=rtpmap:18 G729/8000 a=rtpmap:18 G729/8000
If the answerer supports G.729.1, it will keep the payload type 98 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. 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 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 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 using G.729 (the payload format for G.729 is defined in Section
4.5.6 of RFC 3551 [4]). 4.5.6 of RFC 3551 [4]).
Note that when used at 8 kbps in G.729-compatible mode, the 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 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 advertised (by default, annexb=yes for G729 media type; see
4.1.9 of RFC 3555 [7]). Section 4.1.9 of RFC 3555 [7]).
o The "maxbitrate" parameter is bi-directional. If the offerer sets o The "maxbitrate" parameter is bi-directional. If the offerer sets
a maxbitrate value, the answerer MUST reply with a smaller or a maxbitrate value, the answerer MUST reply with a smaller or
equal value. The actual maximum bit rate for the session will be equal value. The actual maximum bit rate for the session will be
the minimum. the minimum.
o If the received value for "maxbitrate" is between 8000 and 32000 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 but not in the permissible values set, it SHOULD be read as the
closest lower valid value. If the received value is lower than closest lower valid value. If the received value is lower than
8000 or greater than 32000, the session MUST be rejected. 8000 or greater than 32000, the session MUST be rejected.
o The "mbs" parameter is not symmetric. Values in the offer and the o The "mbs" parameter is not symmetric. Values in the offer and the
answer are independent and take into account local constraints. answer are independent and take into account local constraints.
One party MUST NOT start sending frames at a bit rate higher than One party MUST NOT start sending frames at a bit rate higher than
the "mbs" of the other party. The parameter allows to announce the "mbs" of the other party. The parameter allows announcing
this value, prior to the sending of any packet, to avoid the this value, prior to the sending of any packet, to prevent the
remote sender to exceed the MBS at the beginning of the session. remote sender from exceeding the MBS at the beginning of the
session.
o If the received value for "mbs" is greater or equal to 8000 but 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 not in the permissible values set, it SHOULD be read as the
closest lower valid value. If the received value is lower than closest lower valid value. If the received value is lower than
8000, the session MUST be rejected. 8000, the session MUST be rejected.
o The parameters "ptime" and "maxptime" will in most cases not o The parameters "ptime" and "maxptime" will in most cases not
affect interoperability. The SDP offer-answer handling of the affect interoperability. The SDP offer-answer handling of the
"ptime" parameter is described in RFC 3264 [8]. The "maxptime" "ptime" parameter is described in RFC 3264 [8]. The "maxptime"
parameter MUST be handled in the same way. parameter MUST be handled in the same way.
o Any unkown parameter in an offer MUST be ignored by the receiver, o Any unknown parameter in an offer MUST be ignored by the receiver
and MUST NOT be included in the answer. and MUST NOT be included in the answer.
Some special rules apply for mono-directional traffic: Some special rules apply for mono-directional traffic:
o For sendonly streams, the "mbs" parameter is useless and SHOULD o For sendonly streams, the "mbs" parameter is useless and SHOULD
NOT be used. NOT be used.
o For recvonly streams, the "mbs" parameter is the only way to o For recvonly streams, the "mbs" parameter is the only way to
communicate the MBS to the sender, since there is no RTP stream communicate the MBS to the sender, since there is no RTP stream
towards it. So to request a bit rate change, the receiver will 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. need to use an out-of-band mechanism, like a SIP RE-INVITE.
Some special rules apply for multicast: Some special rules apply for multicast:
o The "mbs" parameter MUST NOT be used. o The "mbs" parameter MUST NOT be used.
o The "maxbitrate" parameter becomes declarative and MUST NOT be o The "maxbitrate" parameter becomes declarative and MUST NOT be
negotiated. This parameter is fixed, and a participant MUST use negotiated. This parameter is fixed, and a participant MUST use
the configuration that is provided for the session. the configuration that is provided for the session.
6.2.2. Declarative SDP considerations 6.2.2. Declarative SDP Considerations
For declarative use of SDP such as in SAP [10] and RTSP [11], the For declarative use of SDP such as in SAP [10] and RTSP [11], the
following considerations apply: following considerations apply:
o The "mbs" parameter MUST NOT be used. o The "mbs" parameter MUST NOT be used.
o The "maxbitrate" parameter is declarative and provide the o The "maxbitrate" parameter is declarative and provides the
parameter that SHALL be used when receiving and/or sending the parameter that SHALL be used when receiving and/or sending the
configured stream. configured stream.
7. Congestion control 7. Congestion Control
Congestion control for RTP SHALL be used in accordance with RFC 3550 Congestion control for RTP SHALL be used in accordance with RFC 3550
[3] and any appropriate profile (for example, RFC 3551 [4]). The [3] and any appropriate profile (for example, RFC 3551 [4]). The
embedded and variable bit rates capability of G.729.1 provides a embedded and variable bit rates capability of G.729.1 provides a
mechanism that may help to control congestion, see Section 3 for more mechanism that may help to control congestion; see Section 3 for more
details. details.
The number of frames encapsulated in each RTP payload influences the The number of frames encapsulated in each RTP payload influences the
overall bandwidth of the RTP stream, due to the header overhead. overall bandwidth of the RTP stream, due to the header overhead.
Packing more frames in each RTP payload can reduce the number of Packing more frames in each RTP payload can reduce the number of
packets sent and hence the header overhead, at the expense of packets sent and hence the header overhead, at the expense of
increased delay and reduced error robustness. increased delay and reduced error robustness.
8. Security considerations 8. Security Considerations
RTP packets using the payload format defined in this specification RTP packets using the payload format defined in this specification
are subject to the general security considerations discussed in the are subject to the general security considerations discussed in the
RTP specification [3] and any appropriate profile (for example, RFC RTP specification [3] and any appropriate profile (for example, RFC
3551 [4]). 3551 [4]).
As this format transports encoded speech/audio, the main security As this format transports encoded speech/audio, the main security
issues include confidentiality, integrity protection, and issues include confidentiality, integrity protection, and
authentication of the speech/audio itself. The payload format itself authentication of the speech/audio itself. The payload format itself
does not have any built-in security mechanisms. Any suitable does not have any built-in security mechanisms. Any suitable
external mechanisms, such as SRTP [12], MAY be used. external mechanisms, such as SRTP [12], MAY be used.
This payload format and the G.729.1 encoding do not exhibit any This payload format and the G.729.1 encoding do not exhibit any
significant non-uniformity in the receiver-end computational load and significant non-uniformity in the receiver-end computational load and
thus in unlikely to pose a denial-of-service threat due to the thus are unlikely to pose a denial-of-service threat due to the
receipt of pathological datagrams. receipt of pathological datagrams.
9. IANA considerations 9. IANA Considerations
It is requested that one new media subtype (audio/G7291) is IANA has registered audio/G7291 as a media subtype; see Section 6.1.
registered by IANA, see Section 6.1.
10. References 10. References
10.1. Normative references 10.1. Normative References
[1] International Telecommunications Union, "G.729 based Embedded [1] International Telecommunications Union, "G.729 based Embedded
Variable bit-rate coder: An 8-32 kbit/s scalable wideband coder Variable bit-rate coder: An 8-32 kbit/s scalable wideband coder
bitstream interoperable with G.729", ITU-T Recommendation bitstream interoperable with G.729", ITU-T Recommendation
G.729.1, May 2006. G.729.1, May 2006.
[2] Bradner, S., "Key words for use in RFCs to Indicate Requirement [2] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997. Levels", BCP 14, RFC 2119, March 1997.
[3] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, [3] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
skipping to change at page 13, line 46 skipping to change at page 12, line 45
[6] Freed, N. and J. Klensin, "Media Type Specifications and [6] Freed, N. and J. Klensin, "Media Type Specifications and
Registration Procedures", BCP 13, RFC 4288, December 2005. Registration Procedures", BCP 13, RFC 4288, December 2005.
[7] Casner, S. and P. Hoschka, "MIME Type Registration of RTP [7] Casner, S. and P. Hoschka, "MIME Type Registration of RTP
Payload Formats", RFC 3555, July 2003. Payload Formats", RFC 3555, July 2003.
[8] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with [8] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
Session Description Protocol (SDP)", RFC 3264, June 2002. Session Description Protocol (SDP)", RFC 3264, June 2002.
10.2. Informative references 10.2. Informative References
[9] International Telecommunications Union, "Coding of speech at 8 [9] International Telecommunications Union, "Coding of speech at 8
kbit/s using conjugate-structure algebraic-code-excited linear- kbit/s using conjugate-structure algebraic-code-excited linear-
prediction (CS-ACELP)", ITU-T Recommendation G.729, March 1996. prediction (CS-ACELP)", ITU-T Recommendation G.729, March 1996.
[10] Handley, M., Perkins, C., and E. Whelan, "Session Announcement [10] Handley, M., Perkins, C., and E. Whelan, "Session Announcement
Protocol", RFC 2974, October 2000. Protocol", RFC 2974, October 2000.
[11] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time Streaming [11] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time Streaming
Protocol (RTSP)", RFC 2326, April 1998. Protocol (RTSP)", RFC 2326, April 1998.
skipping to change at page 14, line 24 skipping to change at page 13, line 21
Author's Address Author's Address
Aurelien Sollaud Aurelien Sollaud
France Telecom France Telecom
2 avenue Pierre Marzin 2 avenue Pierre Marzin
Lannion Cedex 22307 Lannion Cedex 22307
France France
Phone: +33 2 96 05 15 06 Phone: +33 2 96 05 15 06
Email: aurelien.sollaud@orange-ft.com EMail: aurelien.sollaud@orange-ftgroup.com
Full Copyright Statement Full Copyright Statement
Copyright (C) The Internet Society (2006). Copyright (C) The Internet Society (2006).
This document is subject to the rights, licenses and restrictions This document is subject to the rights, licenses and restrictions
contained in BCP 78, and except as set forth therein, the authors contained in BCP 78, and except as set forth therein, the authors
retain all their rights. retain all their rights.
This document and the information contained herein are provided on an This document and the information contained herein are provided on an
skipping to change at page 15, line 45 skipping to change at page 14, line 45
such proprietary rights by implementers or users of this such proprietary rights by implementers or users of this
specification can be obtained from the IETF on-line IPR repository at specification can be obtained from the IETF on-line IPR repository at
http://www.ietf.org/ipr. http://www.ietf.org/ipr.
The IETF invites any interested party to bring to its attention any The IETF invites any interested party to bring to its attention any
copyrights, patents or patent applications, or other proprietary copyrights, patents or patent applications, or other proprietary
rights that may cover technology that may be required to implement rights that may cover technology that may be required to implement
this standard. Please address the information to the IETF at this standard. Please address the information to the IETF at
ietf-ipr@ietf.org. ietf-ipr@ietf.org.
Acknowledgment Acknowledgement
Funding for the RFC Editor function is provided by the IETF Funding for the RFC Editor function is provided by the IETF
Administrative Support Activity (IASA). Administrative Support Activity (IASA).
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