Hello Asterisk Community,
We are using Asterisk version 11.25.3 in our CTI system, and we’re experiencing call quality issues We are currently experiencing issues with call quality, specifically intermittent audio and poor sound quality, on our CTI system running Asterisk version 11.25.3. To address these issues, we have made some adjustments to our Jitter Buffer configuration, but we would like to seek advice on optimizing the settings further.
Here is our current configuration:
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
jbenable=yes
jbforce=yes
jbmaxsize=200
jbresyncthreshold=1000
jbimpl=adaptive
jblog=yes
Could you please provide recommendations on the following points?
- Is the
jbmaxsize
setting of 200ms sufficient, or should we consider increasing it to 300ms or 400ms for better handling of network jitter? - For
jbresyncthreshold
, would lowering the threshold to 500ms or 250ms improve audio quality, or is it best left at the default of 1000ms? - Are there any other adjustments or best practices for Jitter Buffer configuration in Asterisk 11.25.3 that we should consider?
Any advice or suggestions from those with experience dealing with similar issues would be greatly appreciated.
Thank you in advance for your help!