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VOIP Codecs and Compression

VOIP Codecs and Compression

Compression software (called a codec) encodes the voice signals into digital data that it compresses into lighter packets that are then transported over the Internet. At the destination, these packets are decompressed and given their original size and converted back to analog voice again, so that the user can hear. Codecs are not only used for compression, but also for encoding, which, simply said is the translation of analog voice into digital data that can be transmitted over IP networks.

VoIP codecs supported by VoIPInvite


G.711 is an ITU-T standard for audio companding. It is primarily used in telephony. The standard was released for usage in 1972. G.711 is a standard to represent 8 bit compressed pulse code modulation (PCM) samples for signals of voice frequencies, sampled at the rate of 8000 samples/second. G.711 encoder will create a 64 kbit/s bitstream requiring a minimum of 128 kbit/s of bandwidth for a single 2 way communication channel.


G.729 is an audio data compression algorithm for voice that compresses voice audio in chunks of 10 milliseconds. DTMF or fax tones in this method use G.711 or out-of-band methods to transport these signals. G.729 is mostly used in Voice over IP (VoIP) applications for its low bandwidth requirement. Standard G.729 operates at 8 kbit/s, but there are extensions, which provide also 6.4 kbit/s and 11.8 kbit/s rates. Also very common is G.729a which is compatible with G.729, but requires less computation.