Chapter 8. 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 mostif not allof 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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