Overview

Chapter 1, "Introduction," mentions the fact that Voice over IP (VoIP) can take advantage of a variety of efficient speech-coding techniques and Chapter 2, "Transporting Voice by Using IP," delves into some detail regarding how digitally encoded speech is carried in an Internet Protocol (IP) network by using the Real-Time Transport Protocol (RTP). Chapter 2 also makes reference to various RTP payload types, which in turn refer to various speech-coding standards. If it were not already obvious, the discussions in these previous chapters should indicate that speech-coding techniques are of particular significance in VoIP networks. In fact, speech coding is a very important topic.

Though some analog systems still remain, they are rare. These days, voice is digitally encoded and shipped around the network (any network) as a stream of 1's and 0's. Speech/voice coding is simply the process by which a digital stream of 1's and 0's is made to represent an analog voice waveform. The process involves converting the incoming analog voice pattern into a digital stream and converting that digital stream back to an analog voice pattern at the ultimate destination. One of the reasons for implementing VoIP is the opportunity to take advantage of efficient voice coding, where fewer bits are used to represent the voice being transmitted, thereby reducing bandwidth requirements and reducing cost.

The initial conclusion could be that one should implement the most bandwidth-efficient coding scheme possible, thereby saving the most money. Unfortunately, however, there is no such thing...