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Codec = Compression/decompression
A codec implements an algorithm (like G.711) to encode information (e.g. the data representing the audio stream) before transmitting it over the network. Some codecs are also designed to reduce the amount of data (compression) required for transporting the information, saving bandwith required on the network.
G.711 is the codec used by ISDN an VoIP. The sampling rate is 64 kbit/s.
If the country code is set to US the audio codec G.711 ulaw is preferred. All other country codes causes the audio codec G.711 alaw to be preferred.
The ITU-T G.722 recommendation specifies speech compression and decompression at rates of 48, 56 or 64 kbps based on Sub-Band Adaptive Differential Pulse Code Modulation (SB-ADPCM).
In the SB-ADPCM technique used, the frequency band is split into two sub-bands (higher and lower) and the signals in each sub-band are encoded using ADPCM.
The G.722 SB-ADPCM audio coder encodes 14 bit and 16 kHz sampled audio signals for transmission over 48, 56 or 64 kbps channels, and provides 7 kHz of audio bandwidth (Wideband audio). The G.722 decoder performs the reverse operation to the encoder.
The G.722 audio codec is used with many applications that require audio frequency bandwidth coding such as video conferencing, speech storing, multimedia, and speaker/microphone digital telephony.
- Coding Rates: 48, 56 or 64 kbps
- Sampling Rate: 16 kHz
- Delay: 125 microseconds
Max. bit rate of 40 kbit/s.
Standard bit rate of 8 kbit/s.