The Video P2P Widget tests the video throughput and quality in a WebRTC peer-to-peer call. This can be used to understand both incoming and outgoing video traffic.
This test connects a WebRTC peer connection from the user’s machine to a cloud browser machine hosted and managed by testRTC. In the video session created, both sides send and receive video.
This gives a rather accurate estimation of the network capabilities and quality to serve video calls.
This can be used as a good indicator in the case of receive only scenarios, where the user isn’t expected to have a webcam or share his video.
Data we collect and share
| Estimated | Out only | The bandwidth estimation WebRTC has on the outgoing direction. |
| Bitrate | In+Out | The bitrate of the video that is actually sent over the connection in each direction. |
| Round Trip Time | In only | The round trip time reported on incoming traffic. The lower the number the better. |
| Packet Loss | In+Out | Packet loss percentage observed in during the test in each direction. |
Things to notice
- You want round trip time and packet loss to have low values in them.
- Bitrate and Estimated bandwidth should have higher values in them. Note that outgoing bitrate might be low if the content of the video being sent is simple or of low resolution (for example, having the camera’s security cover on).