aging the voice traffic, in terms of switching, multiplexing, provisioning, and servicing. Another key application is the possibility to mix voice and computer data in the same telephone line, which first appeared under the name of ISDN (integrated services digital networks), as the precursor of our current Internet. Appendix G 2-channel audio (2 X 2 bytes) Bytes Figure G1 Schematic representation of standard audio-CD frame. The standard of digital voice for telephony, which was released by ITU-T in 1972, is referred to as G.711. 1 Analog-to-digital (A-D) voice conversion is based on the technique of pulse-code modulation (PCM). 2 It is an uncompressed, lossless code, which converts 8000 samples per second into eight-bit codewords, resulting in a channel rate of 8 X 8000 = 64 kbit/s. The two main sampling algorithms used in PCM are the A-law,3 3 as used in Europe, and the u-law, as used in North America and Japan. My earlier work gives more details about these sampling algorithms and their elaborated variants. 5 This book also describes the digital-voice multiplexing (TDM) standards used by telephone operators, from the early plesiosynchronous digital hierarchy (PDH), to the current synchronous digital hierarchy (SDH or SONET). While the original bit rate has been set to 64 kbit/s, voice channels can also be encoded into 16-32 kbit/s, with the same perceived sound quality, and, using more sophisticated algorithms, even down to some 2 kbit/s, with reasonably acceptable quality. There exist many algorithms to perform digital-voice compression, for instance removing the silences in phone conversation (a 40% bandwidth saving!), or predicting coding of voice patterns. Nowadays, the majority of developed countries use digital telephony, whether based on wireline or wireless network systems. With today's Internet, digital telephony is almost a mere commodity, which a number of operators offer for free, regardless of connection time and reach. Music represents a second major application area of digital coding. The revolu