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

[freeswitch at dalethatcher.com]

Brian,

Thanks for getting back to me.

The SVN revision is 7343.  The host is dedicated hardware (to be honest
I didn't even think it would be ok to run freeswitch on a virtual host).

I'll give the QoS settings a go and see if it clears up the problem.
Could you send me a link to an audio file that is known to play fine?

I will send you the SIP details of my server off list if you have a sec
to see if you can replicate the issue.

thanks,

- Dale

On Sat, Jan 26, 2008 at 11:33:01AM -0800, Brian West wrote:
> Dale,
> 	What SVN rev are you running?  Is this a xen or other virtual  
> instance?  I have done extensive testing in this area and haven't seen  
> this... It could be related to QoS bits.  Have you tried to use  
> iptables to tag your traffic?  Check this link http://wiki.freeswitch.org/wiki/QoS/
> 
> FreeSWITCH doesn't tag any traffic with any QoS bits and Asterisk  
> does... So you might have to use the above URL to cause the traffic to  
> get tagged and prioritized on the ADSL router.
> 
> /b
> 
> 
> On Jan 26, 2008, at 7:29 AM, freeswitch at dalethatcher.com wrote:
> 
> > 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
> >
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> 
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