Switching to VoIP

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Voice Channels

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But with longer packet intervals comes increased lag (see Figure 6-2). The longer the

interval, the longer the lag will be between the time the sound is spoken and the time

it is encoded, transported, decoded, and played back for the listener. An IP packet

isn’t transmitted until it is completely constructed, so a VoIP sound frame can’t

travel across the network until it’s completely encoded. A 30 ms sound frame takes a

third longer to encode than a 20 ms one, and inflicts 10 ms more lag, too. As with all

networked apps, lag is bad. It’s especially bad in VoIP.

Long packet intervals have another drawback: the greater the duration of sound car-

ried by each packet, the greater the chance that a listener will notice a negative effect

on the sound if a packet is dropped due to congestion or a network error. Dropping a

packet carrying 20 ms of sound is almost imperceptible with the G.711 codec, but

dropping a 60 ms packet is quite obtrusive. Since VoIP sound frames are carried in

“unreliable” UDP datagrams, dropped packets aren’t retransmitted. Even if TCP

packets were used instead of UDP, error awareness and retransmission would take so

long that, by the time the retransmitted packet arrived at the receiving phone, it

would be hopelessly out of sequence.

Consider that ...