Voice over Ethernet

Voice over Ethernet
Now, at last, something that we can really get our teeth into! Voice traffic is very unusual, in
that it presents an entirely different set of demands to an Ethernet LAN. We are all used to the
usual pressures from applications—namely bandwidth and delay—but jitter is a new problem
for us. In short, jitter is the variation in the delay experienced by successive packets in a flow.
Voice is a streaming protocol. As the analog signal is encoded and broken down into packets,
each packet is transmitted across the network as a unique entity, and the whole is reassembled
into a stream at the receiving end. Obviously, humans don’t speak in packets, and we take
pauses and breaks at random moments, uttering words when we need to. If the packets received
at the end of the link are delayed too long, or if the delay is too variable for the decoder, then
the voice stream cannot be properly reconstructed.
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Understanding Application Needs
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There is a name for this transmission type:
isochronous
. Derived from the Greek words for
“equal” and “time,” it describes processes that require timing to be coordinated. In other
words, isochronous traffic requires that data flow continuously and at a steady rate in close timing
with the ability of the display mechanism to receive and display the image data. Figure 20.3
demonstrates how jitter can affect the output of the playback buffers.
As you can see, the possibility exists for the playback buffer to be empty, full, or half-full. If
the buffer is half-full, then a smoothly created output audio signal will result in good quality
voice reception.
If the buffer is empty, then no audio signal output can be created; this is not a problem if
the reason for the empty buffer is a genuine lack of transmitted data. But if the buffer is full,
then there may be a problem. First, any new arriving packets will be dropped, and time is
insufficient for them to be retransmitted, so they are lost forever. This, however, may not be
the largest problem, because if the reason for the buffer alternating between full and empty
is a variability in the arrival rate of voice-encapsulated data packets, then the output stream
will be of poor quality.
Jitter is probably of greater significance than simple delay in voice networks, because
(up to a certain limit) delay just means that the receiver has to wait a short time for the words.
But jitter results in poor quality voice reception that may be unacceptable to the listener.
Figure 20.4 illustrates a general design model for multimedia traffic, showing how voice will
be integrated into the IP infrastructure. Obviously, we’re focusing on the campus network,
but you can see clearly that as IP datagrams carry voice into the IP cloud, inconsistencies start
to appear in the delivery process.