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<DIV><FONT face=Arial size=2>Depends on how much latency. The
packetization of voice data (and associated digitizing, transcoding, etc)
introduces some latency. Smaller packet size can reduce this, but at the
expense of needing more packets which eats up more CPU time, etc. Also the
jitter buffer size makes a significant difference. For a PBX (LAN)
application this can be quite small, as network processing is fairly
predictable. For stuff going over the internet it needs to be
larger.</FONT></DIV>
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<DIV><FONT face=Arial size=2>I have a small demo setup I'm experimenting with
that only has a couple of SIP phones. They are in the same room and the
delay is was very annoying. I made the jitter buffers smaller and it
helped. With good echo cancellation and more realistic physical separation
this isn't really a problem.</FONT></DIV>
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<DIV><FONT face=Arial size=2>If it is network based, you will see it on a
ping.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Wilton</FONT></DIV>
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