Add documentation+DCHECK for the packet path from transport to call. An upcoming change will remove PeerConnection's involvement in the RTCP packet path and keep the flow consistent with the RTP packet path. Bug: webrtc:11993 Change-Id: I7ce1bbe8bbc352a49310e2e55b55ca5d8d927935 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217389 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33927}
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.