G.729

G.729 is a royalty-free^[1] narrow-band vocoder-based audio data compression algorithm using a frame length of 10 milliseconds. It is officially described as Coding of speech at 8 kbit/s using code-excited linear prediction speech coding (CS-ACELP).^[2] The wide-band extension of G.729 is called G.729.1, which equals G.729 Annex J.

Because of its low bandwidth requirements, G.729 is mostly used in voice over Internet Protocol (VoIP) applications when bandwidth must be conserved, such as for conference calls. Standard G.729 operates at a bit rate of 8 kbit/s, but extensions provide rates of 6.4 kbit/s (Annex D, F, H, I, C+) and 11.8 kbit/s (Annex E, G, H, I, C+) for worse and better speech quality, respectively.

G.729 has been extended with various features, commonly designated as G.729a and G.729b:

• G729: This is the original codec using a high-complexity algorithm.
• G729A or A annex: This version has a medium complexity, and is compatible with G729. It provides a slightly lower voice quality.
• G729B or B annex: This version extends G729 with silence suppression, and is not compatible with the previous versions.
• G729AB: This version extends G729A with silence suppression, and is only compatible with G729B.

Dual-tone multi-frequency signaling (DTMF), fax transmissions, and high-quality audio cannot be transported reliably with this codec. DTMF requires the use of the named telephony events in the RTP payload for DTMF digits, telephony tones, and telephony signals as specified in RFC 4733.

G.729 annexes

G.729 Annex A

G.729a is a compatible extension of G.729, but requires less computational power. This lower complexity, however, bears the cost of marginally reduced speech quality.

G.729a was developed by a consortium of organizations: France Telecom, Mitsubishi Electric Corporation, Nippon Telegraph and Telephone Corporation (NTT)

The features of G.729a are:

• Sampling frequency 8 kHz/16-bit (80 samples for 10 ms frames)
• Fixed bit rate (8 kbit/s 10 ms frames)
• Fixed frame size (10 bytes for 10 ms frame)
• Algorithmic delay is 15 ms per frame, with 5 ms look-ahead delay
• G.729a is a hybrid speech coder which uses Algebraic Code Excited Linear Prediction (ACELP)
• The complexity of the algorithm is rated at 15, using a relative scale where G.711 is 1 and G.723.1 is 25.
• PSQM testing under ideal conditions yields Mean Opinion Scores of 4.04 for G.729a, compared to 4.45 for G.711 (μ-law)
• PSQM testing under network stress yields Mean Opinion Scores of 3.51 for G.729a, compared to 4.13 for G.711 (μ-law)

Some VoIP phones incorrectly use the description "G729a/8000" in SDP (e.g. this affects some Cisco and Linksys phones). This is incorrect as G729a is an alternative method of encoding the audio, but still generates data decodable by either G729 or G729a - i.e. there is no difference in terms of codec negotiation. Since the SDP RFC allows static payload types to be overridden by the textual rtpmap description this can cause problems calling from these phones to endpoints adherring to the RFC unless the codec is renamed in their settings since they will not recognise 'G729a' as 'G729' without a specific workaround in place for the bug.

G.729 Annex B

G.729 has been extended in Annex B (G.729b) which provides a silence compression method that enables a voice activity detection (VAD) module. It is used to detect voice activity in the signal. It also includes a discontinuous transmission (DTX) module which decides on updating the background noise parameters for non speech (noisy frames). It uses 2-byte Silence Insertion Descriptor (SID) frames transmitted to initiate comfort noise generation (CNG). If transmission is stopped, and the link goes quiet because of no speech, the receiving side might assume that the link has been cut. By inserting comfort noise, analog hiss is simulated digitally during silence to assure the receiver that the link is active and operational.

Other extensions

Recently, G.729 has been extended (with Annex J) to provide support for wideband speech and audio coding, i.e., the transmitted acoustic frequency range is extended to 50 Hz - 7 kHz. The respective extension to G.729 is referred to as G.729.1. The G.729.1 codec is hierarchically organized: Its bit rate and the obtained quality are adjustable by simple bitstream truncation.

Licensing

As of January 1, 2017 the patent terms of most licensed patents under the G.729 Consortium have expired, the remaining unexpired patents are usable on a royalty-free basis.^[4] Thus, G.729 can be used free-of-charge. G.729 includes patents from several companies which were until the expiry licensed by Sipro Lab Telecom, the authorized Intellectual Property Licensing Administrator for G.729 technology and patent pool.^[5][6][7][8]

Past patent litigation

AIM IP, a non-practicing entity based in Mission Viejo, CA has made complaints for infringement of U.S. Patent No. 5,920,853 to a number of companies .^[9][10] The patent contains lookup tables including the sequence 1486, 2168, 3751... which is integral to g.729.^[11]

The European counterpart of this patent, according to Google patent records appears to have been withdrawn.^[12]

RTP payload type

G.729 is assigned the static payload type 18 for RTP by IANA.^[13] The rtpmap parameter description for this payload type is "G729/8000".

Both G.729a and G.729b use the same rtpmap description as G.729. G.729a and G.729b are indicated using annexb=no or annexb=yes, respectively. G.729 Annex B (G.729b) is the default in absence of parameter annexb in the Session Description Protocol.^[14]

See also

• List of codecs
• Comparison of audio coding formats
• RTP audio video profile

References

External links

• ITU-T Recommendation G.729 - technical specification
• G.729 Error Recovery for Internet Telephony
• ITU Patent database
• Sipro Lab Telecom (administers the patent pools for G.723.1 and G.729)
• Voiceage's G.729 implementation