Does sipML provide any info about call quality? Something like dropped packets or packets arriving out of order?
I have looked at sipML API documentation, but did not find anything relevant. Also looked into the Developer Tools of Firefox/Chrome, but didn't find such metrics there.
We would like to implement a call quality indicator similar to what other communication tools like Skype for Business or Teams have.
There doesn't seem to be an official API but if you can get a hold of the underlying RTCPeerConnection object (here?) you can use the full getStats() API