WebRTC video stream quality monitoring in Javascript

Question

I use WebRTC for streaming video between peers, but changes in network conditions for some clients often lead to qualitative changes in the resulting video stream. People blame the service for these drops of quality, and it is clear that I (the service) can not do anything about their network conditions. But showing that quality that has fallen due to client-side network conditions is likely to alleviate this problem.

I’ve searched Google and Stackoverflow for some time now and haven’t seen any issues related to the quality detection of incoming audio or video stream. Is there a way to control the quality (current bitrate or dropped frames, whatever) during the broadcast?

Answer 1

The getStats () API is what you are looking for if you want to programmatically receive real-time information. webrtc-internals is a standalone webpage that provides you with more information than getstats as it has access to chrome's internal components, but ultimately most of this data will be accessible to getstats so people can access it from their application .