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| Basic Standards for Audio Coding |
IP telephony also starts with the analogue human voice signal. This signal is coded and
compressed with an analogue-digital converter. Together with a compression algorithm for
bandwidth-efficient data transport, this forms the encoder part of the digital signal transmitter.
In the next step, dedicated portions of the bit stream are formed into Internet Protocol
packets. This is called "packetizing". These packets are then sent through the IP network to
their final destination. The packets can be routed to the receiver in different ways. This might
lead to a varying delay in packet reception. At the receiving end a Jitter Buffer handles the
packets initially and sends on a constant stream of packets in the original sequence for
further processing.
Delayed or missing packets can severely decrease speech quality, although the Jitter Buffer
is able to compensate to a certain extent. Finally the speech packets are re-converted into an
audible analogue signal.
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