As the title suggests, I'm looking for a way to notify users of a webrtc video conference that the quality issues they might be experiencing is due to either network bandwidth or CPU usage or else..
I'm aware of the WebRTC Stats API (getStats) but i'm just really not sure of a proper way of detecting issues. I see I can get access to the bitrate, packet loss, RTT but i'm not really sure of an algorithm to determine video quality based on those metrics.
I know webrtc automatically reduces the resolution based on bandwidth or cpu etc and thought detecting this might be a possibility? Any help is greatly appreciated!
If you're ok with Chrome-proprietary statistics, googCpuLimitedResolution and googBandwidthLimitedResolution as described here might be useful.