Audio drop out on caller site after running pjsua several hours

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Hi Michael,

are you using dsound.c or DirectSound from PortAudio? We normally use
PortAudio with WMME backend, so haven't used DS/PA-DS at all recently. Any
chance you may be able to retest with PA-WMME? (the pjsip "factory setting"
should use PA-WMME by default).

thanks
 Benny


On Thu, May 14, 2009 at 8:25 AM, Michael <michael_zurich at yahoo.com> wrote:

>
> Hi Benny
>
> I think there is a buffer bug in pjmedia capture that makes sound problem
> after several hours.
>
> Use pjsua-i386-Win32 sample using Directsound and set
> snd_auto_close_time = -1 // disable
> Then make a call to activate sound.
> Wait 36 hours.
> Call again from pjsua-i386-Win32 sample. You hear on other site drops.
> It seams that the Directsound capture buffer will make problems after
> several hours running.
>
> Regards
>
> Michael.
>
> --- On Sun, 5/3/09, Michael <michael_zurich at yahoo.com> wrote:
>
> > From: Michael <michael_zurich@xxxxxxxxx>
> > Subject:Audio drop out on caller site after running pjsua several
> hours
> > To: pjsip at lists.pjsip.org
> > Date: Sunday, May 3, 2009, 5:19 PM
> > Hi
> >
> > It seams there is a problem in pjsua (pjsip/pjmedia) with
> > buffers.
> > I have a simple application based on pjsua (1.1 trunk) on
> > Win32.
> > When starting the application I have great sound over
> > Asterisk 1.2 and G711.
> > After several hours. The application is idle, no calls.
> > When a call arrives the caller has bad quality. Drops on
> > audio. Sounds like an empty buffer problem.
> >
> > Do you know this problem? Have you ever let an application
> > idle for 24 hours and the call to this application?
> >
> > Regards
> >
> > Michael.
> >
> >
> >
> >
>
>
>
>
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