December 9, 2023

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

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

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


Some information of G.729 and G.711 in VoIP


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. 


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



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.711 (PCM method: PCM = Pulse Code Modulation: Pulse Code Modulation)
  • Sampling rate: 8kHz
  • Information volume: 64kbps/channel
  • Theoretical delay: 0.125msec
  • Quality: MOS value 4.10


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.

  • G.729 (CS-ACELP method: Conjugate Structure Algebraic Code Excited Linear Prediction)
  • Sampling rate: 8kHz
  • Information volume: 8kbps/channel
  • Frame length: 10msec
  • Theoretical delay: 15msec
  • Quality: MOS value 3.9




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

G.711 and G.729 are two common audio codecs used in Voice over IP (VoIP) communications.

G.711 is an uncompressed audio codec that uses pulse code modulation (PCM) to encode audio. It provides high-quality audio with minimal latency and is best suited for high-bandwidth connections.

However, it requires a large amount of bandwidth and can consume a significant amount of network resources. It is commonly used in situations where audio quality is critical, such as in call centers or business environments.


G.729 is a compressed audio codec that uses a more efficient algorithm to encode audio.

It provides lower bandwidth usage and reduced network resource consumption, making it ideal for low-bandwidth connections. However, it sacrifices some audio quality to achieve this efficiency.

G.729 is commonly used in situations where bandwidth is limited, such as in remote offices or over cellular networks.


Overall, the choice between G.711 and G.729 will depend on the specific needs and constraints of the VoIP system, such as available bandwidth, network resources, and desired audio quality.




Comparison: G.711 vs. G.729


G.711 and G.729 are both popular codecs used in VoIP telephony, but they differ in several aspects. Here are some comparisons:

  1. Bitrate: G.711 has a bitrate of 64 kbps, while G.729 has a lower bitrate of 8 kbps. This means that G.729 can compress audio signals more efficiently than G.711, resulting in smaller file sizes and less bandwidth consumption.

  2. Quality: G.711 is a uncompressed codec, so it provides high quality audio with minimal loss of data. G.729, on the other hand, is a compressed codec and provides lower quality audio, although it is still acceptable for most users.

  3. Compatibility: G.711 is more widely used and supported than G.729. This is because G.711 is a standard codec used in traditional PSTN phone systems, while G.729 is a proprietary codec that requires a license to use.

  4. Delay: G.729 has a lower delay than G.711, which means that there is less latency in the audio signal when using G.729.

  5. Bandwidth: G.729 is more bandwidth-efficient than G.711, which means that it can handle more calls using the same amount of bandwidth. This is why G.729 is often used in low-bandwidth environments, such as mobile networks or satellite connections.


In summary, G.711 provides higher quality audio but consumes more bandwidth, while G.729 provides lower quality audio but consumes less bandwidth.

The choice between the two codecs depends on the specific needs of the application and the available network bandwidth.


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