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This document updates the Real-time Transport Protocol (RTP) payload format to be used for the International Telecommunication Union (ITU-T) Recommendation G.729.1 audio codec. It adds Discontinuous Transmission (DTX) support to the RFC 4749 specification, in a backward-compatible way. An updated media type registration is included for this payload format.
1.
Introduction
2.
Background
3.
RTP Header Usage
4.
Payload Format
5.
Payload Format Parameters
5.1.
Media Type Registration Update
5.2.
Mapping to SDP Parameters
5.2.1.
DTX Offer-Answer Model Considerations
5.2.2.
DTX Declarative SDP Considerations
6.
Congestion Control
7.
Security Considerations
8.
IANA Considerations
9.
References
9.1.
Normative References
9.2.
Informative References
§
Author's Address
§
Intellectual Property and Copyright Statements
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The International Telecommunication Union (ITU-T) Recommendation G.729.1 [ITU‑G.729.1] (International Telecommunications Union, “G.729 based Embedded Variable bit-rate coder: An 8-32 kbit/s scalable wideband coder bitstream interoperable with G.729,” May 2006.) is a scalable and wideband extension of the Recommendation G.729 [ITU‑G.729] (International Telecommunications Union, “Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear-prediction (CS-ACELP),” March 1996.) audio codec. [RFC4749] (Sollaud, A., “RTP Payload Format for the G.729.1 Audio Codec,” October 2006.) specifies the payload format for packetization of G.729.1 encoded audio signals into the Real-time Transport Protocol (RTP) [RFC3550] (Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, “RTP: A Transport Protocol for Real-Time Applications,” July 2003.).
The Annex C of G.729.1 [ITU‑G.729.1‑C] (International Telecommunications Union, “G.729.1 DTX/CNG scheme,” May 2008.) adds Discontinuous Transmission (DTX) support to G.729.1. This document updates the RTP payload format to allow usage of this Annex.
Only changes or additions to [RFC4749] (Sollaud, A., “RTP Payload Format for the G.729.1 Audio Codec,” October 2006.) will be described in the following sections.
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 [RFC2119] (Bradner, S., “Key words for use in RFCs to Indicate Requirement Levels,” March 1997.).
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G.729.1 supports Discontinuous Transmission (DTX), a.k.a. "silence suppression". It means that the coder includes a Voice Activity Detection (VAD) algorithm, to determine if an audio frame contains silence or actual audio. During silence periods, the coder may significantly decrease the transmitted bit rate by sending a small frame called Silence Insertion Descriptor (SID), and then stop transmission. The receiver's decoder will generate comfort noise (CNG) according to the parameters contained in the SID. This DTX/CNG scheme is specified in [ITU‑G.729.1‑C] (International Telecommunications Union, “G.729.1 DTX/CNG scheme,” May 2008.).
G.729.1 SID has an embedded structure. The core SID is the same as the legacy G.729 SID [ITU‑G.729‑B] (International Telecommunications Union, “A silence compression scheme for G.729 optimized for terminals conforming to Recommendation V.70,” October 1996.). A first enhancement layer adds some parameters for narrowband comfort noise, while a second enhancement layer adds wideband information. G.729.1 SID can be 2, 3, or 6 octets long.
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The fields of the RTP header must be used as described in [RFC4749] (Sollaud, A., “RTP Payload Format for the G.729.1 Audio Codec,” October 2006.), except for the Marker (M) bit.
If DTX is used, the first packet of a talkspurt, that is, the first packet after a silence period during which packets have not been transmitted contiguously, MUST be distinguished by setting the M bit in the RTP data header to one. The M bit in all other packets MUST be set to zero. The beginning of a talkspurt MAY be used to adjust the playout delay to reflect changing network delays.
If DTX is not used, the M bit MUST be set to zero in all packets.
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The payload format is the same as in [RFC4749] (Sollaud, A., “RTP Payload Format for the G.729.1 Audio Codec,” October 2006.), with the option to add a SID at the end.
So the complete payload consists of a payload header of 1 octet (MBS and FT fields), followed by zero or more consecutive audio frames at the same bit rate, followed by zero or one SID.
Note that it is consistent with the payload format of G.729 described in section 4.5.6 of [RFC3551] (Schulzrinne, H. and S. Casner, “RTP Profile for Audio and Video Conferences with Minimal Control,” July 2003.).
To be able to transport a SID alone, that is, without actual audio frames, we assign the FT value 14 to SID. The actual SID size (2, 3, or 6 octets) is inferred from the payload size: it is the size of what is left after the payload header.
When a SID is appended to actual audio frames, the FT value remains the one describing the encoding rate of the audio frames. Since the SID is much smaller than any other frame, it can be easily detected at the receiver side, and it will not hinder the calculation of the number of frames. The actual SID size is inferred from the payload size: it is the size of what is left after the audio frames.
The full FT table is given for convenience:
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 |
12-13 | (reserved) | - |
14 | SID | 2, 3, or 6 octets |
15 | NO_DATA | 0 |
DTX has no impact on the MBS definition and use.
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Parameters defined in [RFC4749] (Sollaud, A., “RTP Payload Format for the G.729.1 Audio Codec,” October 2006.) are not modified. We add a new optional parameter to configure DTX.
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We add a new optional parameter to the audio/G7291 media subtype:
- dtx:
- indicates that discontinuous transmission (DTX) is used or preferred. Permissible values are 0 and 1. 0 means no DTX. 1 means DTX support, as described in Annex C of ITU-T Recommendation G.729.1. 0 is implied if this parameter is omitted.
When DTX is turned off, the RTP payload MUST NOT contain SID, and the FT value 14 MUST NOT be used.
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The information carried in the media type specification has a specific mapping to fields in the Session Description Protocol (SDP) [RFC4566] (Handley, M., Jacobson, V., and C. Perkins, “SDP: Session Description Protocol,” July 2006.), which is commonly used to describe RTP sessions.
The mapping described in [RFC4749] (Sollaud, A., “RTP Payload Format for the G.729.1 Audio Codec,” October 2006.) remains unchanged.
The "dtx" parameter goes in the SDP "a=fmtp" attribute.
Some example partial SDP session descriptions utilizing G.729.1 encodings follow.
Example 1: default parameters (DTX off)
m=audio 57586 RTP/AVP 96
a=rtpmap:96 G7291/16000
Example 2: recommended packet duration of 40 ms (=2 frames), maximum bit rate is 20 kbps, DTX supported and preferred.
m=audio 49987 RTP/AVP 97
a=rtpmap:97 G7291/16000
a=fmtp:97 maxbitrate=20000; dtx=1
a=ptime:40
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The offer-answer model considerations of [RFC4749] (Sollaud, A., “RTP Payload Format for the G.729.1 Audio Codec,” October 2006.) fully apply. In this section we only define the management of the "dtx" parameter.
The "dtx" parameter concerns both sending and receiving, so both sides of a bi-directional session MUST have the same DTX setting. If one party indicates it does not support DTX, DTX must be deactivated both ways. In other words, DTX is actually activated if, and only if, "dtx=1" in the offer and in the answer.
A special rule applies for multicast: the "dtx" 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.
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The "dtx" parameter is declarative and provides the parameter that SHALL be used when receiving and/or sending the configured stream.
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The congestion control considerations of [RFC4749] (Sollaud, A., “RTP Payload Format for the G.729.1 Audio Codec,” October 2006.) apply.
The use of DTX can help congestion control by reducing the number of transmitted RTP packets and the average bandwidth of audio streams.
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The security considerations of [RFC4749] (Sollaud, A., “RTP Payload Format for the G.729.1 Audio Codec,” October 2006.) apply.
DTX introduces no new security issue. The ability to interrupt transmission during silence periods is a well-known feature in RTP.
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It is requested that one new parameter for media subtype (audio/G7291) is registered by IANA, see Section 5.1 (Media Type Registration Update).
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[ITU-G.729.1] | 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. |
[ITU-G.729.1-C] | International Telecommunications Union, “G.729.1 DTX/CNG scheme,” ITU-T Recommendation G.729.1 Annex C, May 2008. |
[RFC2119] | Bradner, S., “Key words for use in RFCs to Indicate Requirement Levels,” BCP 14, RFC 2119, March 1997 (TXT). |
[RFC4749] | Sollaud, A., “RTP Payload Format for the G.729.1 Audio Codec,” RFC 4749, October 2006 (TXT). |
[RFC3550] | Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, “RTP: A Transport Protocol for Real-Time Applications,” STD 64, RFC 3550, July 2003 (TXT). |
[RFC4566] | Handley, M., Jacobson, V., and C. Perkins, “SDP: Session Description Protocol,” RFC 4566, July 2006 (TXT). |
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[ITU-G.729] | 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. |
[ITU-G.729-B] | International Telecommunications Union, “A silence compression scheme for G.729 optimized for terminals conforming to Recommendation V.70,” ITU-T Recommendation G.729 Annex B, October 1996. |
[RFC3551] | Schulzrinne, H. and S. Casner, “RTP Profile for Audio and Video Conferences with Minimal Control,” STD 65, RFC 3551, July 2003 (TXT). |
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Aurelien Sollaud | |
France Telecom | |
2 avenue Pierre Marzin | |
Lannion Cedex 22307 | |
France | |
Phone: | +33 2 96 05 15 06 |
Email: | aurelien.sollaud@orange-ftgroup.com |
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