I am doing 200 concurrent calls. As per requirement I am recording only few calls using java AMI mix monitor. channel.startMonitoring(channel.getId(), “wav”, true);
Calls getting recorded successfully but the duration of the calls getting increased. The calls which I am recording have huge fluctuation/disturbance in voice as well.
Note :- Calls which are not getting recorded are perfectly fine.
I check server using top command. CPU and memory looks fine. We tried after server restarted as well.
This is my first question on community. Its my humble request please ignore if I have missed any protocol. Can somebody help plz.
Do you mean latency (the delay between the speaker speaking and the listener hearing), rather than duration?
Also, as this sounds like some sort of system overload, you need to describe the system on which it is running.
A contributory factor can be transcoding. MixMonitor recording has to be done with a linear time domain signal. If you are using complex codecs (which tend to work more in the frequency domain, un-recorded the media may just be copied through, but to record it, the encoding needs to be undone, which can involve significant CPU processing. For that reason, you also need to tell us about the codecs in use.
David, there is no latency actually. We are broadcasting audio/mp3 in every call. When somebody answered the call the voice totally breaks.
Example:- suppose we uploaded an audio of 30 seconds. If call is not recorded then billable_seconds remains 30. On contrast if call is recorded then billable_seconds become around 50 seconds.
we are using 4 core processor, 32 GB RAM server.
For codec, how can i check which codec is used on my server?
thanks
Look at your configuration to see which ones you offered to use; this is a significant design decision for VoIP. Use the core show channel CLI command to see what was actually selected.