How to debug webRTC internal dump

WebRTC is a powerful technology that enables real-time communication of audio, video and data in web browsers and native apps. However, it can also be a complex and challenging domain to debug when things go wrong. In this blog post, I will show you how to use the WebRTC internal dump feature to collect and analyze useful information about the state and behavior of a WebRTC session.

The WebRTC internal dump feature allows you to capture a detailed log of events and messages exchanged by the WebRTC components during a session. This log can be very helpful to identify and diagnose issues such as connection failures, poor media quality, signaling errors, codec mismatches and more. The internal dump can also be used to generate statistics and graphs that can help you understand the performance and quality of your WebRTC application.

To enable the WebRTC internal dump feature, you need to set a flag in your browser or app that tells WebRTC to save the log to a file. The flag is different depending on the platform you are using. For example, on Chrome you can use the --enable-logging --vmodule=/webrtc/=2 command line arguments to enable verbose logging of WebRTC modules and save the log to a file named chrome_debug.log in your user data directory. On Android, you can use the org.webrtc.Logging.enableLogToDebugOutput(Logging.Severity.LS_INFO) method in your app code to enable logging of WebRTC modules to the Android logcat. You can then use the adb logcat -v threadtime | grep -F "adb shell ps | grep org.webrtc | awk '{print $2}'" command to filter and save the log to a file.

Once you have enabled the WebRTC internal dump feature and collected the log file, you can use the webrtc-internals-analysis tool to parse and analyze the log. This tool is a Python script that can be found in the tools_webrtc directory of the WebRTC source code repository. You can run the tool with the following command: python webrtc-internals-analysis.py --log_file=path/to/log/file --output_dir=path/to/output/dir. The tool will generate a JSON file that contains a structured representation of the events and messages in the log, as well as an HTML file that displays various statistics and graphs based on the JSON data.

The HTML file generated by the webrtc-internals-analysis tool provides a comprehensive overview of your WebRTC session, including information such as:

• The signaling state and messages exchanged by the peer connection

• The ICE candidates and connectivity checks performed by the ICE agent

• The DTLS handshake and SRTP encryption parameters used by the transport layer

• The RTP packets sent and received by each media track

• The codec parameters and payload types used by each media track

• The RTCP feedback messages sent and received by each media track

• The bandwidth estimation and congestion control algorithms used by each media track

• The media quality metrics such as packet loss, jitter, delay, bitrate, frame rate, resolution, etc.

• The CPU and memory usage of the WebRTC components

By inspecting these statistics and graphs, you can gain valuable insights into how your WebRTC session works and what factors affect its quality and performance. You can also compare different sessions or scenarios to identify potential problems or areas for improvement.

The WebRTC internal dump feature is a powerful tool that can help you debug and optimize your WebRTC applications. However, it is not without limitations. Some of the limitations are:

• The internal dump feature may incur some overhead and affect the performance of your WebRTC session, especially if you enable verbose logging or collect large amounts of data.

• The internal dump feature may not capture all the information or events that are relevant to your specific use case or issue. For example, it may not capture network conditions, browser settings, user actions, etc.

• The internal dump feature may not be compatible with all platforms or versions of WebRTC. For example, some platforms may not support logging to a file or may use different formats or flags for logging.

• The webrtc-internals-analysis tool may not support all features or formats of WebRTC. For example, it may not support new codecs, extensions, protocols, etc.

Therefore, you should use the WebRTC internal dump feature with caution and complement it with other debugging tools and methods as needed.

I hope this blog post has given you a useful introduction to how to use the WebRTC internal dump feature to debug your WebRTC applications. If you have any questions or feedback, please feel free to leave a comment below or contact me at [e-mail]. Thank you for reading!