Analysing WebRTC dumps using chrome webrtc-internals
Oppdatert: 28. jul.
WebRTC is an open-source project that enables real-time communication between web browsers and mobile applications via simple APIs. Testing WebRTC video calls is essential to ensure a good user experience. Chrome WebRTC dumps provide a useful tool for analyzing and testing the quality of WebRTC video calls.
Here are the steps to test a WebRTC video call using Chrome WebRTC dumps:
1. Install the Chrome WebRTC Internals extension(It's an old extension last updated on 21 Dec 2021 ): The first step is to install the Chrome WebRTC Internals extension from the Chrome Web Store. This extension provides access to the Chrome WebRTC debugging and diagnostic tools.
2. Enable the Chrome WebRTC diagnostic logging: To enable the Chrome WebRTC diagnostic logging, type chrome://webrtc-internals in the Chrome address bar and press Enter. Click on the "Start" button next to "Event Log." This will enable the diagnostic logging for all WebRTC events in Chrome.
3. Record the WebRTC dump: Once you have enabled the diagnostic logging, start the WebRTC video call and let it run for a few minutes. After the call, stop the diagnostic logging by clicking on the "Stop" button next to "Event Log." This will generate a WebRTC dump that contains all the data packets transmitted during the call.
4. Analyze the dump: Once you have the dump, you can analyze it using the Chrome WebRTC diagnostic tools. Click on the "PeerConnection" tab to see detailed information about the WebRTC connection. Look for packet loss, jitter, and latency, which are the key metrics that impact call quality.
5. Calculate metrics: Calculate the key metrics, including Mean Opinion Score (MOS), packet loss rate, jitter, and delay. MOS is a subjective quality metric that rates call quality on a scale of 1 to 5, with 5 being the best. Packet loss rate measures the percentage of data packets lost during the call. Jitter measures the variability in the delay of data packets. Delay measures the time it takes for data packets to travel from the sender to the receiver.
6. Interpret the results: Interpret the results and identify any issues that impact call quality. For example, high packet loss, jitter, or latency can result in poor call quality, including choppy video or delays. Use the results to optimize the network, troubleshoot issues, and improve call quality.
7. Repeat the test: Repeat the test to ensure that the changes you made have improved call quality. Test the call under different network conditions, such as high traffic or low bandwidth, to ensure that it works well in all scenarios.
Chrome WebRTC dumps provide a useful tool for testing WebRTC video calls. By enabling diagnostic logging and analyzing the dump, you can identify issues that impact call quality and optimize the network to improve the user experience. Repeat the test under different network conditions to ensure that the call works well in all scenarios.