For the past couple weeks we have noted choppy outbound audio on voip calls. Affects all three voip trunks outbound. Inbound is clear.
Probably a month or more ago, I changed all our extensions to pjsip without known issue.
Trunks remain SIP.
No hardware changes or known internet issues.
A ping to one affected voip provider from my windows PC gave: 224 packets, 0% lost, min 43ms, max 205ms, average 52ms
Ping from my pbx to one affected voip provider gave: 99 packets transmitted, 99 received, 0% packet loss, time 98125ms
In sip status, it looks like the ping times for the sip trunks are 45ms, 20ms, 175ms. The 45ms trunk is certainly affected.
NAT settings: external address is my WAN address with internet provider. Local networks has my LAN address and subnet.
codecs allowed: ulaw, alaw, g726, gsm, g726
Inbound (callee) audio remains clean.
Calls out to Dahdi trunk are clean.
I changed one extension back to SIP and outbound audio still choppy.
https://pastebin.freepbx.org/view/5fd1642e
Asterisk 14.7.5
FreePBX 14.0.3.6