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Voice quality depends on four main factors, Bandwidth,
Delay, Packet loss and Jitter.
Bandwidth - Usually measured as the number of bits
transferred per second. VoIP requires anywhere from about 24kbits
per second up to about 90kbits per second. Bandwidth is
rarely an issue if you are running Cable, DSL or Fiber.
Delay- Usually measured in milliseconds. Obviously the
longer the delay the more difficult a conversation becomes.
Anything under 100ms is good. The ITU recommend that one way
delay should not exceed 400ms for acceptable speech quality.
Packet Loss - Usually measured as a percentage. Since
your voice is being transmitted digitally over the internet
it will be broken down into segments. These segments become
the packets that are actually transmitted. A packet loss of
up to four or 5% will go largely unnoticed. Anything above
that and call quality will drop.
Jitter - Jitter when applied to VoIP is a variation in packet transit
delay. The causes of jitter are typically queuing,
contention and serialization effects on the path through the
network. Faster, higher bandwidth networks tend to have less
jitter whereas slower networks tend to have more congestion
and more jitter. A QoS device like an ATA can help eliminate
some of this jitter.
All of the above factors are taken into account in the
summary you get at the end of the test.
For an in depth technical view on how
your voice is digitized and transmitted by way of the
internet read our series of
technical blogs. |
