[Freeswitch-users] Audio Quality of recordings

[Scott Ross]

I've developed a tool which leverages freeswitch event_socket to originate
2 calls (sofia/external/number at 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.
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