Switching to VoIP

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Chapter 6: Replacing the Voice Circuit with VoIP

Ethernet isn’t the only data link suitable for carrying VoIP packets—ATM, frame-

relay, point-to-point circuits, and other technologies can be used, and each intro-

duces its own overhead factors.

Decoding and Playback

When a VoIP packet is received, it is decoded according to the codec employed to

encode it. It is then played back on the analog hardware of the receiving endpoint—a

speaker—while undergoing DAC, or digital-to-analog conversion. Decoding

generally takes about as much processing power as encoding, depending on the

codec employed.

Figure 6-4. An Ethernet-encapsulated 20 ms VoIP packet

Table 6-2. VoIP codec bandwidth consumption

Codec Encoded sound bandwidth Ethernet overhead bandwidth Total bandwidth

G.711 64 kbps 31.2 kbps 95.2 kbps

G.726 32 kbps 31.2 kbps 63.2 kbps

G.728 16 kbps 31.2 kbps 78.4 kbps

a

a

G.728 uses four voice frames at 16 kbps per packet. This accounts for the deviation in overhead bandwidth.

G.729A 8 kbps 31.2 kbps 39.2 kbps

GSM 13 kbps 31.2 kbps 44.2 kbps

IP

Header

IP

Payload

UDP

Header

UDP

Payload

RTP

Header

RTP

Payload

G.711

Payload

160 bits

64 bits

96 bits

1,280 bits

Ethernet

Header

Ethernet

Header

128 bits

Ethernet CRC and Bumper

176 bits

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Voice Channels

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Most IP phones and ATAs support several codecs, as shown in Table 6-3. All sup-

port G.711 using both the

µlaw and Alaw scales, and a majority support G.729A,

though with variance in quality and completeness of their implementation. It’s fair to

say that G.711 and G.729A are the two most popular VoIP codecs in use today.

Things that degrade playback quality

Several factors can degrade the quality of audio transmitted over the network:

Jitter

This effect occurs when gaps between packets occur at durations greater than

the packet interval. The effect is missing or garbled speech. Jitter can be caused

by network congestion or processing overloads on the encoding or decoding

endpoints (this usually doesn’t happen on dedicated devices, only on soft-

phones). Some IP phones and softPBXs offer a jitter buffer to compensate for

packets arriving out of sequence or at odd intervals, but jitter buffers introduce

lag—and lag is bad.

G.729A and Asterisk

Digium’s implementation of the G.729A codec for Asterisk is a licensed commercial

version made by Voice Age (http://www.voiceage.com). The codec is patent-protected,

so if you want to use G.729A endpoints with the Asterisk server, you must pay for a

commercial license. Digium sells the licensed, and GPL-friendly, version of G.729A at

the price of around $10 per simultaneous call. Using unlicensed versions of G.729A

(which do exist) violates the GPL under which Asterisk is distributed, because the GPL

requires end users to adhere to local patent law, by which G.792A is governed in sev-

eral nations.

Table 6-3. Codecs supported by some leading VoIP endpoint devices

Phone / ATA G.711 G.726 G.728 GSM Speex G.729A

Cisco 7960 IP Phone Yes No No No No Yes

Avaya 4602 IP Phone Yes No No No No Yes

Grandstream Budgetone 101 IP Phone Yes Yes Yes No No Yes

Digium IAXy i100 ATA Yes Yes No No No No

Grandstream Handytone ATA Yes Yes Yes No No Yes

Cisco ATA-186 ATA Yes No No No No Yes

3com 3102 IP Phone Yes Yes No No No Yes

SNOM 220 IP Phone Yes No No Yes No Yes

X-Lite Softphone Yes No No Yes Yes No