Question

I'm using WebRTC to stream video between peers, but changes in network conditions for some clients often produce quality changes in the received video stream. People blame the service for these quality drops and clearly I (the service) cannot do anything about their network conditions. But showing an indication that the quality dropped due to network conditions on the client's end would most likely alleviate this problem.

I've been searching Google & Stackoverflow for a while now and haven't seen any questions related to quality detection of incoming audio or video stream. Is there a way to monitor the quality (current bitrate or dropped frames, anything) during the live stream?

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Solution

The getStats() API is what you are looking for if you want to programmatically access information in real time. webrtc-internals is a separate webpage, that is providing you more informations than getstats because it has access to chrome internals, but eventually most of those info will be made available to getstats so people can have access to them from within their app.

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