System information

APPENDIX B
Protocols for VoIP
The Internet is a telephone system that’s gotten uppity.
—Clifford Stoll
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 under-
standing the differences between the various codecs and protocols will give you a
greater appreciation and understanding of the system as a whole.
This appendix explains Voice over IP and what makes VoIP networks different from
the traditional circuit-switched voice networks that were the topic of Appendix A. 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 (some more briefly than others):
• 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 connection can carry is directly related to the type of
codec implemented. We’ll also explore the differences between the following codecs
in regard to bandwidth requirements (compression level) and quality:
• G.711
• G.726
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