since it makes no sense for the WebRTC usage of DTLS and increases the size of the last handshake flight considerably Guarded by killswitch WebRTC-DisableTlsSessionTicketKillswitch BUG=webrtc:367181089 Co-authored-by: Jody Ho <jodyho@meta.com> Change-Id: I4bb17bba8a17c65c8e0fefe2d8962974703feee7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362526 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: David Benjamin <davidben@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#43046}
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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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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