The Video Bandwidth Widget tests the video throughput bandwidth estimation of WebRTC. This can be used to understand how much UDP traffic WebRTC believes it can use for a video session.
This test connects a WebRTC peer connection via the TURN servers of the tested infrastructure, sending video through the connection and checking what WebRTC estimates as the available bandwidth.
This gives a good approximation of what you should expect to effectively have available in a video call session for the outgoing video. It should be taken into account here that the estimates are based on the browser and are done over a short period of time – available bandwidth changes dynamically.
Data we collect and share
| Bandwidth Estimate | The bandwidth estimation WebRTC has on the outgoing direction. |
| Jitter | How stable the connection is. The lower the number the better. |
| Round Trip Time | The round trip time reported. The lower the number the better. |
| Packet Loss | Packet loss percentage observed during the test. |
Things to notice
You want jitter, round trip time and packet loss to have low values to them.
The bandwidth estimate isn’t what we send over the network, but rather what WebRTC believes it ‘can’ send over the network.