Refactor RTP header extension ID allocation to be persistent and shared. - Introduced RtpHeaderExtensionRecorder and RtpHeaderExtensionPicker in call/payload_type_picker. - Added SuggestRtpHeaderExtensionId and AddRtpHeaderExtensionMapping to PayloadTypeSuggester interface. - Implemented persistent RTP header extension ID mapping in SdpPayloadTypeSuggester. - Refactored pc/media_session.cc to use PayloadTypeSuggester for all RTP header extension ID allocations, replacing UsedRtpHeaderExtensionIds. - Ensured shared mapping between local and remote extensions for the same transport. - Maintained RFC 8285 compliance for one-byte and two-byte ID allocation. - Updated tests to explicitly allow mixed extensions when two-byte IDs are expected. - Removed obsolete UsedRtpHeaderExtensionIds and its unit test. Bug: webrtc:503013383 Change-Id: Ic98522b1eec52c736a73bd034493f19ac6524b5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/465300 Auto-Submit: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#47492}
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.