I am using gvoice per the config and directions on the wiki: http://wiki.freeswitch.org/wiki/Google_Voice We're using a mixed bag of hardware SIP capable phones, linksys and cisco mostly. flow is: google voice <> ipcop firewall / NAT <> dmz'ed freeswitch box(running current) <> bunch of SIP phones the freeswitch server is a dell poweredge, couple xeons at 3.something, four gig of ram, couple sata2 hard drives in raid1. load stays under 14% basically all the time. iostat shows very low usage for basically everything, our pipe is comcast, 20/8. we have, seemingly randomly, audio quality problems. most phone calls are crisp and clear, and one in ten, (Sometimes in streaks of 3-4 calls) we'll have choppy sound both directions. "Hello my name is mike and i'm calling for" sounds like: "llo ame is and alling" this seems to be per-call, while one person in the office is having poor call quality most others are fine at the same time. i have enabled QoS, and prioritized traffic to and from the freeswitch box. our pipe is not saturated during the problems. i've played with VAD, send-silence, and rtp-timer. i've tried changing the codec the phones use with no real effect, i've tried changing the codec used in jingle but we're unable to dial out using anything but PCMU. Where should i look next? Thanks in advance <3 Trel Nadal