Chapter 8: Protocols for VoIP

The telecommunications industry spans over 100 years, and Asterisk integrates most - if not all - of the major technologies that it has made use of over the last century. To make the most out of Asterisk, you need not be a professional in all areas, but understanding the differences between the various codecs and protocols will give you a greater appreciation and understanding of the system as a whole.

This chapter explains Voice over IP and what makes VoIP networks different from the traditional circuit-switched voice networks that were the topic of the last chapter. We will explore the need for VoIP protocols, outlining the history and potential future of each. We'll also look at security considerations and these protocols' abilities to work within topologies such as Network Address Translation (NAT). The following VoIP protocols will be discussed:

  • IAX
  • SIP
  • H.323
  • MGCP
  • Skinny/SCCP
  • UNISTIM

Codecs are the means by which analog voice can be converted to a digital signal and carried across the Internet. Bandwidth at any location is finite, and the number of simultaneous conversations any particular connection can carry is directly related to the type of codec implemented. In this chapter, we'll also explore the differences between the following codecs in regards to bandwidth requirements (compression level) and quality:

  • G.711
  • G.726
  • G.723.1
  • G.729A
  • GSM
  • iLBC
  • Speex
  • MP3

We will then conclude the chapter with a discussion of how voice traffic can be routed reliably, what causes echo and how to minimize it, and how Asterisk controls the authentication of inbound and outbound calls.

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