[Freeswitch-users] Audio Quality of recordings

[Scott Ross]

Also:
have tried running on both xen vm and physical hosts. a=ptime:20  was there
on FS siptrace and verified with wireshark capture - no apparent timing
issue.

Just wanted to post here in case there is something else obvious I
overlooked before posting on Jira.

Thanks


On Fri, Nov 8, 2013 at 3:28 PM, Scott Ross <slross16 at gmail.com> wrote:

> 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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