How can I improve the quality of playback regular files and music on hold? Sometimes the sound is not smooth (choppy). I use the alaw & ulaw file formats. The asterisk version is 16.3, 4GB of RAM, linux 64bit. Is this amount of memory enough for smooth playback? Now the number of simultaneous calls is slow (up to 5). Calls are recorded (does recording seriously affect performance)?
Make sure you have a timing source configured.
Find and remove the resource bottleneck in your specific system.
Could you expand, please, how I can configure it for pjsip and find problems? I cannot find, how to do this. I am using PJSIP, no hardware equipment, just a sip connection.
The timing source is set at a global level, by installing a suitable timing module. https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces
However, I think this is more likely to to be a resource issue.
If using a virtual machine, try a physical machine until you have it working.
My asterisk is installed on a physical machine. Now it has two timing modules loaded: res_timing_pthread and res_timing_timerfd.
I think I have figured it out. It is important to set Quality of Service for the trunk. I am using FreePBX which sets these parameters automatically only for regular extensions (phones), but not for trunk.
That means that:
you have a network congestion problem;
your routers actually honour quality of service markings.
This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.