[Freeswitch-users] Audio rate changes and clipping on playback

[freeswitch at dalethatcher.com]

Hello all!

I'm experiencing some problems with clipping and the rate of playback of
wav files from Freeswitch.  I've not raised this before as I thought the
problems were due to hosting over an ADSL line.  However I've now seen
the same issues running from a dedicated host.

My test setup is:

	Client <-> NAT ADSL <-> Freeswitch

I've replicated using both the Gizmo and FWD clients from Windows.  If I
substitute Asterisk with Freeswitch then the audio comes over fine.  The
two test samples are one of myself and the fpm-sunshine.wav that is
bundled with asterisk.

I've tried playing around with the codecs but see the same problems
regardless.  Also I've performed a test with a conference hosted on the
Freeswitch instance with myself and friend dialling in using Gizmo and
the audio is fine.  Which implied to me that the link between Gizmo to
Freeswitch is ok.

So I'm guessing that there is something wrong with the samples that is
causing the problem?  They're both in this format:

$ file fpm-sunshine.wav
RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz

My deployment platform is Debian stable and the Freeswitch SVN revision
is 7343.  I've tried looking in Jira but keep getting a system error
page (for search terms 'audio', 'wav' and others).

Is anyone else seeing the same issues or has a solution?

thanks,

- Dale