Refactor RTP Header Extension Management into RtpTransport Moves the caching, validation, and history tracking of RTP header extensions from BaseChannel to RtpTransport. This aligns with the BUNDLE model and simplifies the concurrency model by handling this on the network thread. Bug: webrtc:42222117 Change-Id: Iae4c65c938ee085d242cc3a27b5abaff5d7471de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455460 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#47275}
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.
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.