Real time communications can be challenging: browsers, network, hardware and operating systems need to successfully perform complex tasks for a call to be successful.
Understanding what goes wrong, often on the far end, is challenging for developers.
test.webrtc.org captures media from the microphone and camera at various resolutions and sends it back to itself through a TURN relay server, all encrypted.
Once the media is back, Javascript code looks for various issues such as black video frames and silent audio as well as testing network capabilities.
If an issue is discovered, users can easily download or upload the report. When the latter is chosen, a unique URL is created, which can be used when filing issues.
We encourage developers to take the code from github and integrate similar functionality in their own UX. This should be part of any "support request" flow. We also encourage developers to contribute!
Real time communications can be challenging: browsers, network, hardware and operating systems need to successfully perform complex tasks for a call to be successful.
Understanding what goes wrong, often on the far end, is challenging for developers.
test.webrtc.org captures media from the microphone and camera at various resolutions and sends it back to itself through a TURN relay server, all encrypted.
Once the media is back, Javascript code looks for various issues such as black video frames and silent audio as well as testing network capabilities.
If an issue is discovered, users can easily download or upload the report. When the latter is chosen, a unique URL is created, which can be used when filing issues.
We encourage developers to take the code from github and integrate similar functionality in their own UX. This should be part of any "support request" flow. We also encourage developers to contribute!
https://github.com/webrtc/testrtc