If one wants to monitor connection quality from the client side, I think the major factor other than disconnections is "adaptive bitrate (ABR)". I'm pretty sure that the getStats browser api is one of the best ways to do this, but the word "adaptive" doesn't appear in
https://www.w3.org/TR/webrtc-stats/
I can see in Chrome72/Win10 that "candidate-pair" events have property "availableOutgoingBitrate" as well as "bytesReceived" and "bytesSent", and that event "outbound-rtp" has properties "bytesSent" and "packetsSent", and that event "transport" has "bytesSent" and "bytesReceived". I'd guess that "candidate-pair" events appear only during call setup.
Has a best practice emerged about how to use these events and properties over the duration of a call (assuming we probe at an interval of 150msec to 1sec) to monitor adaptive bitrate?