Choppy audio

Hey all,

Running FreePBX as a VM currently provisioned with 2gb ram and 1 vCPU. Our connection is average (100mbit/5mbit) and we get a ~20ms ping to our sip provider (sipstation.)

We are running Digium D50s in the office (10) with the latest firmware etc…

We seem to be having issues with choppy audio (cut in/out) with just enough regularity to be an annoyance. Bought and installed the .729 codec but didn’t seem to help (assuming it was installed correctly.)

Im a bit of a neophyte with respects to the PBX world but is there a best practices approach as to where to start when hunting these issues down?


The obvious weak link in that configuration is the use of a virtual machine. If you need to use a VM you need access to an expert on configuring the host, to ensure that Asterisk has a sufficiently low scheduling latency and smooth timestamping. Only allowing it one virtual CPU suggests that not much thought has been given to the tuning of the VM host.

You should be aiming for a scheduling latency of less than about 10ms, and similar short term clock accuracy, although, at a push, you might get away with up to about 50ms.

Performance is going to depend on what else is sharing the host.

I’ve read m as M, g as G, and b as B where that makes more sense.

Even for a small system, a VM shares resources with the host. I’d allot at least 2 CPU (cores), 4 GB RAM, and run the VM on a SSD using the Fixed Size option, and a Bridged NIC.