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 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
