Computer Telephony Encyclopedia

The ITU-T Recommendation that s become the standard for converting analog toll-quality voice lines into toll-quality digital ones. G.711 is also used extensively in newer forms of IP telephony. G.711 is a Pulse Code Modulation (PCM) scheme operating at a 8 kHz sample rate, with 8 bits per sample. Since a signal must be sampled at twice its highest frequency (as dictated by the Nyquist theorem), G.711 can thus encode frequencies between 0 and 4 kHz. The algorithm allows for the transmission and reception of A-law and u-law voice by converting linear Pulse Code Modulation (PCM) input signals (13 bits for the international A-law standard and 14 bits for u-Law) sampled at an 8 kHz sampling rate into an 8-bit compressed floating-point PCM representation. The actual technique of converting between linear PCM and G.711 PCM is known as companding (compressing/ expanding). The analog-speech waveform, once having been encoded as binary words is then transmitted serially, at digital bit rates of 48, 56, or 64 Kbps. ISDN channels and digital phone sets on digital PBXs use G.711. Support for this algorithm is required for ITU-T compliant videoconferencing (the H.320 / H.323 standard).
Although considered to be a popular codec or vocoder, G.711 excessive bandwidth output at 64 kbps is generally considered to be pretty much uncompressed, and is thus used as a reference against which the speech quality of voice compression algorithms with high compression and lower bit rates are measured.
When the H.323 IP interoperability standard ruled the...