I've developed a tool which leverages freeswitch event_socket to originate 2 calls (sofia/external/number@ip) whom then navigate an IVR, connect to the same conference bridge then take turns playing back a .wav while the partner call records channel input. The goal is to run the two recorded files against a PESQ library to monitor whether our conference bridges are performing as expected. Problem: I'm hearing poor channel quality (static, degradation) from the recordings and when I listen in on the calls while the test is running. My PESQ scores are coming out much lower than expected. Out of curiosity I tried asterisk and scores were about a point higher (2.9 vs 3.9) so I should be able to rule out the hardware/environment/destination lines I'm running it on. I need help improving freeswitch audio quality for my use-case. I know this is asked alot - I've read all the threads/wiki articles I can find and tried: loading only necessary modules, running a single codec (ulaw), I've tried ensuring my playback reference is also ulaw format without headers, I've tried running on several different OSs: Ubuntu 11.04, RHEL 6.8, RHEL 5.6. Even originating on two separate sofia profiles (although I don't think this should have mattered regardless). I prefer freeswitch over asterisk in all other aspects but need to solve this (important) quality issue. Let me know what outputs I can provide.