Delay, Jitter and Lost Packets

One of the big targets on VoIP has been delay. When VoIP first became credible as a telephone technology it was easy to attack the fact that it simply could not be as good as legacy telephone service because of the inherent delays in processing of voice. That was true then and it is true today, but for human conversations it is fine, generally.

The first point about delay has to do with the encoding and various steps needed to transmit voice over IP. This delay depends on the processing power and older PCs suffered some but this is not a significant issue today. But overall delay can be caused by other factors.

One fundamental of transmission of voice or any signal is the speed of light. This first became very noticeable with satellite communications, and the question was often how many "hops" (between satellites) were involved. On the Internet we can similarly question what route was taken since packets can be routed circuitously over the Internet and be delayed. No matter what, half way around the world is a long way.

The assembly of an audio stream (your voice) requires that the data packets arrive to be processed recreating the original transmitted voice. When this occurs perfectly, the quality is high indeed. Because we have packets routed independently, we don't know that they always arrive in sequence and occasionally a packet will be lost. This is a significant VoIP issue where the transmission is over the Internet. To have the highest quality voice we want to wait to get all packets, but we cannot wait since this a real conversation, hence delays do cause a deterioration of voice quality. A buffer (jitter buffer) is employed to collect the incoming packets before decoding the output. Designers work to trade-off the voice quality versus the delay to getting the best quality voice -and a useful conversation.

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