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.
| 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 | 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.
