VoIP codec: What is the difference betwwen G.729 d and G.711?

Depend on VoIP gateway equipment, there are 5 standards for voice compression settings in the setting interface, namely G.711-uLaw, G.711-aLaw, G.723-53k, G.723-63k , G729, G.711 and G.729 are two very popular codes in enterprise VoIP deployment. 

G.711 sound quality is superior compared to voice streams encoded via G.729. G.711 is usually used in LAN environments where there is no bandwidth problem, and the bandwidth requirement is about 80kbps, including overhead bandwidth.

 G.729 is usually used in WAN environments with limited bandwidth, and the bandwidth requirement is about 30kbps.

G.711 is a nonlinear quantization of voice analog signals, and there are two subdivisions: G.711 A-law and G.711 u-law. 

Different countries and places will choose one as their own standard. G.711 bitrate is 64kbps. Detailed information can be found in relevant specs in the ITU, and some performance parameters are listed below:

G.723.1 is a dual-rate speech encoder, a compression algorithm recommended by ITU-T for voice or other audio signals in low-rate multimedia services;

its target application systems include multimedia communication systems such as H.323 and H.324 , the algorithm has become one of the necessary algorithms in the IP telephone system;

the frame length of the encoder is 30ms, and there is a look-ahead of 7.5ms, and the algorithm delay of the encoder is 37.5ms; the encoder first performs traditional The filtering of the telephone bandwidth (based on G.712), and then the speech signal is sampled at a traditional 8000-Hz rate (based on G.711), and converted into 16-bit linear PCM code as the input of the encoder.

The output is reversed in ××× to reconstruct the speech signal; the high-rate encoder uses multi-pulse maximum likelihood quantization (MP-MLQ), and the low-rate encoder uses the Algebraic Code Excited Linear Prediction (ACELP) method, the encoder Both and ××× must support the two rates, and be able to convert the two rates between frames. 

This system is also capable of compressing and decompressing music and other audio signals, but it is optimized for speech signals; silence compression is used which performs discontinuous transmission, which means that during silence periods the bitstream adds man-made noise. 

In addition to reserving bandwidth, this technique keeps the modem of the transmitter in continuous operation and avoids the on and off of the carrier signal.

The algorithm adopted by G.729 is Conjugate Algebraic Code Excited Linear Prediction (CSACELP), which is based on the CELP coding model; it can achieve very high voice quality (long-distance voice quality) and very low algorithm delay; the algorithm frame length 10ms, the encoder includes 5ms look-ahead, and the algorithm delay is 15ms;

its reconstructed voice quality is equivalent to 32kb/s ADPCM (G.726) in most working environments, and the MOS score is greater than 4.0; input 16bitPCM voice signal when encoding, and output Binary bit stream; input as binary bit stream during decoding, and output 16bitPCM voice signal;

on the basis of voice signal 8KHz sampling, encode after 16bit linear PCM, and the data rate after compression is 8Kbps; it has the equivalent of 16: 1 compression ratio.

The G.729 series is widely used in the current VOIP, and there are many related branches. You can directly get the source code and related documents from the ITU website.