Audio Codec

We are currently using Freepbx ver 12.0.68 with the default audio codec. It works great most of the time. Other times I have calls randomly drop or the audio completely drops off or other times I have intermitted audio drops for like 1 sec.

Do I need to enable Jitter Buffer and set this up. Or do I need to buy the g.729 codec from Digium and install it. Some of my remote sites that VPN back into this PBX server have limited bandwith but even the places with high bandwidth experience this sometimes. I don’t very high usage on the systems resources either.

This is not a codec issue, this is a network issue.

A random 1-second drop normally wouldn’t be an issue, but when you’re using VoIP, it stands out.

well we run a WAN, but Drops happen whether or not I’m on the same subnet as the server or at remote WAN site over the VPN. What would be the next best course of action on this since it is a network issue.

1.) isolate the PBX to its own VLAN / subnet?
2.) isolate the PBX to its own subnet and create isolated subnets at each WAN site and use different vpn tunnels to get back to the PBX.?