Audio drop out on caller site after running pjsua several hours

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Hi Benny

I compiled pj with portaudio PA-DS instead of factory PA-WMME and reduced latency to 40ms. Latency is the reason for PA-DS. This works fine for some hours. 

Kind Regards

Michael

On 14.05.2009, at 22:07, Benny Prijono <bennylp at teluu.com> wrote:

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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