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 ining audio or video stream. Is there a way to monitor the quality (current bitrate or dropped frames, anything) during the live stream?

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 ining audio or video stream. Is there a way to monitor the quality (current bitrate or dropped frames, anything) during the live stream?

• javascript
• video-streaming
• html5-video
• webrtc

• 1 If they are using Chrome, you can use chrome://webrtc-internals/. I answered a similar question on how to read the data dump you grab from there. – Benjamin Trent Commented Jul 30, 2014 at 12:53

2 Answers 2

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.

You can use webrtc-issue-detector to monitor the quality of webrtc session on the client side. It uses getStats() under the hood and identifies issues that may affect media quality like client CPU issues, network or server side issues.