Analysis FAQ

You can use testRTC to analyze WebRTC dump files taken from your real devices. In order to obtain a webrtc-internals dump file, please perform the following steps: How to download a webrtc-internals dump file In your local b...

When you look at our graphs for a specific test run results of a given probe, what you are actually looking at are specific parameters in getstats on the channels of the session. Here is our mapping: Audio Charts Outgoing an...

Here we provide a video and a step by step guide for that. In general: Open chrome://webrtc-internals in a tab inside Chrome Open your WebRTC application in a separate tab Run your session, but DON’T end it Go to the chrome://webrtc-intern...

testRTC takes into account the main quality metrics that affect voice and video quality. That includes bitrate, packet loss, jitter and delay. It looks at their values as well as their variance over time. Note: We’ve introdu...

You may want to know in a test result if the media was sent directly or relayed via a TURN server. And if it went through a TURN server, you’d like to know if that was done over UDP, TCP, or TLS. For that, if you ...

We are asked from time to time if our platform can assist in catching lip sync issues, which are also known as A/V drift or other similar names. The problem with lip sync testing is that it requires knowing the source media and being...

At testRTC we try hard to simplify the analysis process. We do this by visualizing the results as much as possible and making the important information pop out. While at it, we also make sure you always have access to as much...