I have been trying to troubleshoot issues at our remote office where the phones latency seems to be running high leading to intermittent quality issues during phone calls. I’ve tried everything from replacing the infrastructure (switches, cables) to testing on another FreePBX box, to going through Site-to-Site VPN, and over the public IP. Here are the relevant details:
PBX Firmware: 3.211.63-9
Asterisk (Ver. 220.127.116.11)
SIPStation for trunks (Averages 80ms)
T1 at Main Office dedicated to phone/server traffic (SIP, RTP)
T1 at Branch Office dedicated to phone traffic (SIP, RTP)
Sonicwall NSA 220 at each branch with Site-to-Site VPN
30 Phones (24 at Main Branch, 6 at Remote)
Remote Office Phones are:
(4) Polycom 501
(1) Polycom 601
(1) Polycom Soundstation behind a Linksys PAP2T
Running a ping or traceroute from one office to the other is almost always between 20-30ms. Very rare packet loss after replacing infrastructure.
Main Branch Phone Latency averages 20-40ms, Phone quality is perfect and without issue.
Remote Branch Phone Latency averages 85-125ms, Phone calls experience brief drops of audio, muffled sounds, general jitter issues.
Something very interesting though, the Polycom Soundstation that sits behind the Linksys PAP2T runs right at about 42ms latency on average and works fine. No issues whatsoever with it.
As mentioned above, connecting over the Site-to-Site VPN shows absolutely no improvement over going across the public IP.
This issue has been the bane of my existence for the last few months now and I cannot express how grateful I would be for help in resolving it. Look forward to hearing your ideas! Thank you.
EDIT: Forgot to add, phones are all using G.711
Have considered trying G.729 but felt like something else was at play here besides the codecs.