G.711 is a high bit rate (64 Kbps) ITU standard codec. It is the native language of the modern digital telephone network. Its formal name is Pulse Code Modulation (PCM) of voice frequencies which is a very commonly used waveform codec. Voice and audio signals are analog, whereas data network is digital. The transformation of analog signal to digital signal is done by Analog to Digital Converter (ADC). The process of analog to digital conversion in Pulse Code Modulation is done in three steps; Sampling, Quantizing and Codification. Sampling is the process of encoding an analog signal in digital form by reading (sampling) its level at precisely space intervals of time. Quantization is done by converting the height of the obtained samples to a finite number of discrete values. Encoding happens by means of segmented or continuous way. In continuous encoding, the quantization intervals have different width, from small values which correspond to low level signals, to greater values, corresponding to high level signals. In segmented encoding, he operation range is divided into a finite number of groups. Each interval of the same group has the same width, being different from other groups. Normally segmented encoding is used. There are two main versions of G.711 codec: A-Law and U-Law. U-Law is used in American and Japan PCM systems and A-law is used in the other parts of the world like European PCM systems. Using G.711 for VoIP will give the best voice quality; since it is the same codec used by the PSTN network and ISDN lines, it sounds just like using a regular or ISDN phone. It also has the lowest latency (lag) because there is little to no need for buffering, which costs processing power. The downside is that it takes more bandwidth than other codecs, up to 84 Kbps including all the UDP and IP overhead. However, with increasing broadband bandwidth, this should not be a problem. G.711 is supported by most VoIP providers.

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