srtp: add UseCryptex API to SrtpSession and SrtpTransport Adds the UseCryptex(enable, require, sending_session) API on SrtpSession and UseCryptex(enable, require) on SrtpTransport. Wires libsrtp's srtp_set_stream_use_cryptex through DoSetKey, and adds a require-cryptex post-decrypt check via ParseRtpExtensionProfile (new helper in modules/rtp_rtcp/source/rtp_util) for the cases libsrtp does not yet enforce; see https://github.com/cisco/libsrtp/pull/805. No callers yet; the cryptex SDP negotiation wires this up in a follow-up. Bug: webrtc:455813732 Change-Id: I407400d42912fb303c93c715e7110dafcc3d0cab Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/470600 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/main@{#47707}
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.