Page 222 - Asterisk™: The Future of Telephony

‡
actually hear. The purpose of the various encoding algorithms is to strike a balance
between efficiency and quality. §
Originally, the term codec referred to a COder/DECoder: a device that converts between
analog and digital. Now, the term seems to relate more to COmpression/
DECompression.
Before we dig into the individual codecs, take a look at Table 8-1—it’s a quick reference
that you may want to refer back to.

Table 8-1. Codec quick reference
Codec Data bitrate (Kbps) License required?
G.711 64 Kbps No
G.726 16, 24, 32, or 40 Kbps No
G.729A 8 Kbps Yes (no for passthrough)
GSM 13 Kbps No
iLBC 13.3 Kbps (30-ms frames) or 15.2 Kbps (20-ms frames) No
Speex Variable (between 2.15 and 22.4 Kbps) No

G.711

G.711 is the fundamental codec of the PSTN. In fact, if someone refers to PCM (dis-
cussed in the previous chapter) with respect to a telephone network, you are allowed
to think of G.711. Two companding methods are used: μlaw in North America and
alaw in the rest of the world. Either one delivers an 8-bit word transmitted 8,000 times
per second. If you do the math, you will see that this requires 64,000 bits to be trans-
mitted per second.
Many people will tell you that G.711 is an uncompressed codec. This is not exactly
true, as companding is considered a form of compression. What is true is that G.711
is the base codec from which all of the others are derived.
G.711 imposes minimal (almost zero) load on the CPU.

‡ “Aoccdrnig to rsereach at an Elingsh uinervtisy, it deosn’t mttaer in waht oredr the ltteers in a wrod are, the
olny iprmoetnt tihng is taht frist and lsat ltteres are in the rghit pclae. The rset can be a toatl mses and you
can sitll raed it wouthit a porbelm. Tihs is bcuseae we do not raed ervey lteter by istlef, but the wrod as a
wlohe.” (The source of this quote is unknown―see http://www.bisso.com/ujg_archives/000228.html.) We do
the same thing with sound―if there is enough information, our brain can fill in the gaps.
§ On an audio CD, quality is far more important than saving bandwidth, so the audio is quantized at 16 bits
(times 2, as it’s stereo), with a sampling rate of 44,100 Hz. Considering that the CD was invented in the late
1970s, this was quite impressive stuff back then. The telephone network does not require this level of quality
(and needs to optimize bandwidth), so telephone signals are encoded using 8 bits, at a sampling frequency
of 8,000 Hz.

194 | Chapter 8: Protocols for VoIP