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To test: - ensure openssl files live in /usr/include/openssl - make a config with the following values: ``` rtc_build_ssl = false rtc_ssl_root = "/usr/include/openssl ``` - compile rtc_unittests There will be link errors, but no compile errors. Bug: None Change-Id: I3d28b1613e11a716b11c7def808100ef58ab75f5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358300 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42714}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
Development
See here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation