Log dropped oversized RTCP packets RTCPSender ignores the result of RtcpPacket::Create(), which fails for packets larger than the maximum packet size. Such packets are dropped without any indication, which hid oversized transport feedback at high packet rates, see https://webrtc-review.googlesource.com/c/src/+/514401. Log a warning with the packet size and the maximum packet size when this happens. Bug: webrtc:583184158 Change-Id: Ic5f34886bfc257aeb133282ff4c7f677badb86f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/514500 Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#49193}
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.