[Asterisk-Users] Audio Delay in Meetme

[murray at murraylisook.com]

When conducting a conference call (meetme) with SIP endpoints - Cisco
7960, XLite, and Grandstream sip phones all on the local LAN - we
experience an audio delay of about a half second.  This makes the call
less than business quality, sounding more like a satelite connection
and leading people to talk over eachother.  There is no delay, or
virtually imperceptible delay between the same stations on a station to
station call even when * stays in the audio path. 

Our timing source is a T100P (the only card in the system, and
configured as the primary timing source).  This server was built from a
clean install taken from the CVS on 8/23.  All the stations are using
the G.711 codec.

Is anyone else experiencing this?  Are there adjustments or changes we
could make to decrease latency?

Murray Lisook
Televerde