End user license agreement for g729 implementation read before using: carefully read the terms of this g729 implementation license agreement and appendices, if any ('license') before installing, copying or using the implementation, as defined in this license.

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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 predictionspeech 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. 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:

• G.729: This is the original codec using a high-complexity algorithm.
• G.729A or A annex: This version has a medium complexity, and is compatible with G.729. It provides a slightly lower voice quality.
• G.729B or B annex: This version extends G.729 with silence suppression, and is not compatible with the previous versions.
• G.729AB: This version extends G.729A with silence suppression, and is only compatible with G.729B.
• G.729.1 or J annex: This version extends G.729A and B with scalable variable encoding using hierarchical enhancement layers.

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.

• 1G.729 annexes
• 2Licensing

G.729 annexes[edit]

G.729 Annex A[edit]

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 (80 bits) 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[edit]

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.

G.729 Annex J (G.729.1)[edit]

G.729 Annex J, maintained by G.729.1, provides support for wideband speech and audio. It defines variable bit-rate wideband enhancement using up to 12 hierarchical layers. The core layer is an 8 kbit/s G.729 bitstream, the second layer is a 4 kbit/s narrowband enhancement layer, and the third 2 kibt/s layer is a bandwidth enhancement layer. Further layers provide wideband enhancement in 2 kbit/s steps. The G.729.1 uses three-stage coding: embedded Code-excited linear prediction (CELP) coding of the lower band, parametric coding of the higher band by Time-Domain Bandwidth Extension (TDBWE), and enhancement of the full band by a predictive transform coding by Time-Domain Aliasing Cancellation (TDAC). Bit rate and the obtained quality are adjustable by simple bitstream truncation.

Licensing[edit]

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] 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[edit]

AIM IP LLC, a CaliforniaLimited Liability Company based in Mission Viejo, CA^[9] filed 17 patent infringement lawsuits^[10] in the Central District Courts of California accusing 22 different companies, including, Cisco Systems, Polycom and others of infringing U.S. Patent No. 5,920,853.^[11][12] The '853 patent was filed at the United States Patent and Trademark Office in 1996 by Rockwell International. The inventors listed on the '853 patent are Benyassine Adil, Su Huan-Yu and Shlomot Eyal.^[13]

In 2000, the '853 patent was assigned by Rockwell International to Conexant Systems,^[14] an American-based software developer and fabless semiconductor company, which began as a division of Rockwell before being spun-off as its own public company.^[15] In 2010, the '853 patent was sold by Conexant Systems to AIM IP LLC, a CaliforniaLimited Liability Company based in Mission Viejo.^[14]

The '853 patent contains patent claims which cover lookup tables used in G.729. The patent has since expired and is no longer in force due to its patent term expiring.^[16]

RTP payload type[edit]

G.729 is assigned the static payload type 18 for RTP by IANA.^[17] 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.^[18]

See also[edit]

References[edit]

1. ^Michael Graves (March 6, 2017). 'It's Official! The patents on G.729 have expired'.
2. ^International Telecommunications Union, Standardization Sector (ITU-T), Study Group 15 (1993-1996), Recommendation G.729, March 1996.
3. ^ITU-T (January 2007). 'G.729 : Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear prediction (CS-ACELP)'(PDF): i. Retrieved 2009-07-21.Cite journal requires |journal= (help)
4. ^Sipro Lab Telecom (2017-01-28). 'About G.729'. Archived from the original on 2017-02-02.
5. ^Sipro Lab Telecom Website
6. ^VoiceAge Corporation (2007-10-14). 'G.729 Licensing'. Archived from the original on 2007-10-14. Retrieved 2009-09-17.
7. ^Sipro Lab Telecom (2007-10-25). 'FAQ G.729 and G.723.1'. Archived from the original on 2007-10-25. Retrieved 2009-09-17.
8. ^Sipro Lab Telecom (2006-10-29). 'G.729 IPR Pool'. Archived from the original on 2006-10-29. Retrieved 2009-09-17.
9. ^'Business Search - Results'. Business Search - Business Entities - Business Programs | California Secretary of State.
10. ^'Patent US 5,920,853 A'.
11. ^'RPX Litigation Search'.
12. ^'Aim Ip LLC v. Cisco Systems Inc et. al. patent lawsuit'. Archived from the original on February 1, 2014.
13. ^'United States Patent: 5920853'.
14. ^ ^ab'Assignment abstract of title for Application 08702780'.
15. ^Mark Lapedus (November 10, 1998). 'Rockwell Semi spin-off Conexant will target communications IC market'. EE Times.
16. ^'US5920853A - Signal compression using index mapping technique for the sharing of quantization tables'. Google Patents.
17. ^'Real-Time Transport Protocol (RTP) Parameters'. Iana.org. Retrieved 2013-09-18.
18. ^S. Casner, P. Hoschka (July 2003). 'MIME Type Registration of RTP Payload Formats'. Retrieved 2013-02-27.

External links[edit]

• ITU-T Recommendation G.729 - technical specification.

I am developing Softphone dialer applications for android, iOS etc with PJSIP. I need to know how to integrate G.729 Codec with the project. I've seen articles related to building G.729 Codec using Intel IPP in PJSIP Website itself. But when I contacted Intel Support Team, they told me that it was only supported in older versions of IPP. Now they are not supporting arm processors and G.729 Codec is also not coming along with the sample project. Is there any other way to integrate the codec with PJSIP project.

Thanks in advance

5 Answers

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I had the same requirement for one of my VOIP app. But haven't found any solution or stack overflow answer for a long. So finally i come to know there is another open source 'linphone'(not pjsip stack) integrated with g729codec.

So i took the g729 codec library alone from linphone and created a patch for pjsip. Also I have tested with my iOS app it works well. You can download the patches from my git repo https://github.com/saravanandm/pjsip-g729codec

First you need to download g729 patch from pjsip-android-builder library https://github.com/VoiSmart/pjsip-android-builder/tree/master/g729_patch

Download pjsip , unzip it; put g729_patch folder and pjsip folder into one folder.In g729_patch open in editor install.sh and edit the line 5 (put the correct name of your folder with pjsip), for me it was:PJPROJECT_BASE_FOLDER='../pjproject-2.7.2'

In terminal cd to g729_patch directory and run ./install.sh (you need autoconf utility to be installed for that); patch can be applied manually as well, see instructions in readme.

Create file config_site.h with the following content:

for me it was for android but can be for other platform so 1st line can vary, also other options can be added.Put this file to /pjlib/include/pj.

cd to your pjsip directory and run:

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./configure-android --enable-g729-codec

for me it was android so i ran ./configure-android but may be other.on this step any other configure flags can be applied, for example for me it wasNDK_TOOLCHAIN_VERSION=4.9 TARGET_ABI=armeabi-v7a ./configure-android --use-ndk-cflags --enable-g729-codec

then run:

make dep && make clean && make

That's it - pjsip is compiled with g729 codec.

It seems that PJSIP already support G.729

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Please, follow the instructions here:

Here you found istruction for linux, maybe usefule with some fixes:

For me using IPP would be overkill. AFAIR it is also not free.

Use GSM codec module as a template, modify it and add open source G.729/a (asterisk, freeswitch) implementation as third party library.

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You might have a look at CSipSimple project (https://code.google.com/p/csipsimple/), they use pjsua and G729 codec can be included into the application.

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