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        Summary table   G.711 codec process   Audio samples   New codecs G.711.1 -G722.2--G729.1      

    Codec Summary table

    Voice transmission is analogical, whereas the data network is digital. The process to sample analogical waves into digital information is made by an encoder-decoder (CODEC). There are many standards to sample an analogical voice signal into a digital one. The process is often quite complex. Most of the conversions use pulse code modulation (PCM) or variations

    In addition, the CODEC zip the sequence of data, and sometimes provides echo cancellation. The compression of the waveform can save bandwidth. This is especially interesting in low speed connections so you can have more VoIP connections at the same time. Another way to save bandwidth is using the silence suppression. The goal is not to send packages when there is no voice in the conversations.

    Next is a table with the most known codecs in use:

    - Bit Rate - The rate at which bits are transmitted over a communication path. Normally expressed in Kilobits per second (Kbps)
    - Sampling Rate - the number of samples taken per second when digitizing sound. The quality of the digital reproduction improves as the number of samples taken per second increases.
    - Frame size - The time between packets sent
    - MOS - (Mean Opinion Score). It is a subjective measure of sound quality from 1 to 5.

    In order to understand better the codec process and the parameters expressed in the table we recommended to read the section of G.711 codec process where it is possible to learned how it works the G.711 codec.

    Number Standard by Description Bit rate (kb/s) Sampling rate (kHz) Frame size (ms) Remarks
    MOS (Mean Opinion Score)
    G.711 * ITU-T Pulse code modulation (PCM) 64 8 Sampling U-law (US, Japan) and A-law (Europe) companding
    G.711.1 ITU-T Pulse code modulation (PCM) 80-96 Kbps 8 Sampling Improvement og G.711 to provide an audio bandwidth of 50 Hz to 7 kHz More info
    G.721 ITU-T Adaptive differential pulse code modulation (ADPCM) 32 8 Sampling Now described in G.726; obsolete.
    G.722 ITU-T 7 kHz audio-coding within 64 kbit/s 64 16 Sampling Subband-codec that divides 16 kHz band into two subbands, each coded using ADPCM
    G.722.1 ITU-T Coding at 24 and 32 kbit/s for hands-free operation in systems with low frame loss 24/32 16 20  
    G.722.2 AMR-WB ITU-T Adaptive Multi-Rate Wideband Codec (AMR-WB) 23.85/ 23.05/ 19.85/
    18.25/ 15.85/ 14.25/
    12.65/ 8.85/ 6.6
    16 20 is mainly used for speech compression in the 3rd generation mobile telephony. More info
    G.723 ITU-T Extensions of Recommendation G.721 adaptive differential pulse code modulation to 24 and 40 kbit/s for digital circuit multiplication equipment application 24/40 8 Sampling Superceded by G.726; obsolete. This is a completely different codec than G.723.1
    G.723.1 ITU-T Dual rate speech coder for multimedia communications transmitting at 5.3 and 6.3 kbit/s 5.6/6.3 8 30 Part of H.324 video conferencing. It encodes speech or other audio signals in frames using linear predictive analysis-by-synthesis coding. The excitation signal for the high rate coder is Multipulse Maximum Likelihood Quantization (MP-MLQ) and for the low rate coder is Algebraic-Code-Excited Linear-Prediction (ACELP).
    G.726 ITU-T 40, 32, 24, 16 kbit/s adaptive differential pulse code modulation (ADPCM) 16/24/32/40 8 Sampling ADPCM; replaces G.721 and G.723.
    G.727 ITU-T 5-, 4-, 3- and 2-bit/sample embedded adaptive differential pulse code modulation (ADPCM) var.   Sampling ADPCM. Related to G.726
    G.728 ITU-T Coding of speech at 16 kbit/s using low-delay code excited linear prediction 16 8 2.5 CELP.
    G.729 ** ITU-T Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear-prediction (CS-ACELP) 8 8 10 Low delay (15 ms)
    G.729.1 ITU-T Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear-prediction (CS-ACELP) 8/12/14/16/
    8 10 Improvement og G.711 to provide an audio bandwidth of 50 Hz to 7 kHz More info
    GSM 06.10 ETSI Regular­Pulse Excitation Long­Term Predictor (RPE-LTP) 13 8 22.5 Used for GSM cellular telephony.
    LPC10 USA Government Linear-predictive codec 2.4 8 22.5 10 coefficients.
    Speex     8, 16, 32 2.15-24.6 (NB)
    4-44.2 (WB)
    30 ( NB )
    34 ( WB )
    iLBC     8 13.3 30  
    DoD CELP American Department of Defense (DoD) USA Government   4.8   30  
    EVRC 3GPP2 Enhanced Variable Rate CODEC 9.6/4.8/1.2 8 20 Se usa en redes CDMA
    DVI Interactive Multimedia Association (IMA) DVI4 uses an adaptive delta pulse code modulation (ADPCM) 32 Variable Sampling  
    L16   Uncompressed audio data samples 128 Variable Sampling  
    SILK Skype   From 6 to 40 Variable 20 Harmony codec is related with SILK

    * G711 has two versions called U-law (US, Japan) and A-law (Europe) . U-law is in relation with the T1 standard used in North America and Japan. The A-law is relation with the E1 standard used in the rest of the world. The difference is the method to sample the analog signal. In both schemes, the signal is not sampled linearly, but in a logarithmic way. For more information about the differences you could visit G.711 A Law versus u Law.

    ** There are different versions of g729 codec that it is interesting to explain because this codec is very used nowadays.
    G729: original codec
    G729A or A annex: it is a simplification of G729 and it is compatible with G729. He is less complex but it has less quality.
    G729B or B annex: G729 with silence suppression and not compatible with the previous ones
    G729AB: g729A with silence suppression and only compatible with G729B.
    Besides, every version of G729 have 8Kbps of bitrate but there are versions with 6.4 kbps (D annex) and 11.4 Kbps (E annex).


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