search for: 320ms

...- average 7% --> ping test Latency - average 300ms -> sip show peers - latency ranges from 200-330 but stays within 10ms of initial value on ping test I tried to implement the jitterbuffer in zap. /etc/asterisk/zapata.conf jitterbuffers=16 ; covers the 300ms latency? 20ms each x16 = 320ms On the Polycom IP430's, I setup the jitterbuffer. The audio was still poor to the cell phone user. Jitter Buffer Minimum - 80 Jitter Buffer Shrink - 1000 Jitter Buffer Maximum - 220 I tried these values, but the phone stopped passing ALL audio. Jitter Buffer Minimum - 80 Jitter Buffer Shrink...