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Voice quality network requirements
particularly in the presence of echo. Long WAN transports must be considered as a major
contributor to the network delay budget, averaging approximately 10-20 ms per 1000 miles.
Some transport mechanisms, such as Frame Relay, may add additional delay. Thus, staying
within 150 ms (end to end) may not be possible for all types of connections.
Finally, one-way delay over 400 ms on signaling links between port networks and the S8700
Media Server can cause port network instability.
A network assessment is highly recommended to measure latency (also jitter and packet loss)
and make sure all values are within bounds before implementing IP Telephony.
The following are suggested guidelines for one-way network delay between endpoints, meaning
LAN/WAN measurements not including IP phones:
80 ms (milliseconds) delay or less yields the best quality.
80 ms to 180 ms delay can give Business Communication quality. This is much better than
cell-phone quality and in fact is very well suited for the majority of businesses.
Delays exceeding 180 ms may still be quite acceptable depending on customer
expectations, analog trunks used, codec type, etc.
Again, there is a trade-off between voice quality and the technical and monetary constraints
which businesses confront daily.
Codec delay
Some delay will also be added by various codec algorithms compared to G.711. G.729, for
example, adds:
approximately 10 ms of algorithmic delay in each direction
another 5 ms look ahead
plus signal processing delays.
The compression algorithm in G.723.1 uses multiples of 30 ms samples per packet, which
results in increased latency over codecs configured to use 20 ms or less samples per packet.
Jitter
Jitter is thought of as the statistical average variance in arrival time between packets or
datagrams. To compensate for jitter, many vendors implement a de-jitter buffer in their VoIP
endpoints. The purpose of the jitter buffer is to hold incoming packets for a specified period of
time before forwarding them to the de-packetization (and decompression) process. A jitter
buffer is designed to smooth packet flow. In doing so, it will also add packet delay.
206 Avaya Application Solutions IP Telephony Deployment Guide
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