[pjsip] Audio drop out on caller site after running pjsua several hours
michael_zurich at yahoo.com
Thu May 14 03:25:49 EDT 2009
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.
--- On Sun, 5/3/09, Michael <michael_zurich at yahoo.com> wrote:
> From: Michael <michael_zurich at yahoo.com>
> Subject: [pjsip]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
> It seams there is a problem in pjsua (pjsip/pjmedia) with
> I have a simple application based on pjsua (1.1 trunk) on
> 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?
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