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Voice quality network requirements
246 Avaya Application Solutions IP Telephony Deployment Guide
 
Some of the issues that must be considered when designing a network for VoIP are:
One-way delays in excess of 250 ms can cause the well-known problem of "talk-over." 
This occurs when both parties talk at the same time because the delay prevents them from 
realizing that the other person has already started talking. 
In some applications, delays less than 150 ms can impact the perceived quality, 
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, can 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-series Server can cause port network instability. 
Again, there is a trade-off between voice quality and the technical and monetary constraints 
which businesses confront daily. For this reason, Avaya suggests the following guidelines for 
one-way LAN/WAN delay between endpoints, 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 delay range is 
much better than cell-phone quality if echo is properly controlled and, in fact, is very well 
suited for the majority of businesses.
Delays exceeding 180 ms might still be quite acceptable depending on customer 
expectations, analog trunks used, codec type, and the presence of echo control in 
endpoints or network equipment. 
The Converged Network Analyzer (CNA) system is capable of providing ongoing 
measurements of network delay (see 
will also generate alarms when network delay rises to levels that are detrimental for voice 
quality. For more information on CNA, see 
Codec delay
In addition to delay incurred in the network, codecs in the endpoints also add some delay. The 
delay of the G.711 codec is minimal. However, the G.729 codec, 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 multiple blocks (called frames) of 30 ms voice 
samples per packet. This results in increased latency over codecs configured to use 20 ms or 
less samples per packet.
The G.722 codec adds a 0.82 ms delay.