Consolidate RtpTransport members in JsepTransport. Merges unencrypted_rtp_transport_ and dtls_srtp_transport_ into a single rtp_transport_ member of type std::unique_ptr<RtpTransport>. Updates JsepTransport constructor to accept a single transport. Updates JsepTransportController to instantiate the appropriate transport (encrypted or unencrypted) and pass it. This simplifies JsepTransport by removing mutually exclusive members and aligning with the polymorphic RtpTransport interface. Bug: webrtc:360058654 Change-Id: If7ce6099b4b228d0c07fb0064c1843afbe7b7ea8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/429040 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46341}
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.