2bac7da134
These RTP header extensions are used for Unified Plan SDP / BUNDLE and replace SSRC signaling. Previously, the RTPSender would attach these header extensions to every packet when configured. Now, the header extensions will be attached to every packet until the an RTCP RR is received on that SSRC which indicates the receiver knows what MID/RID the SSRC is associated with. This should reduce overhead by 2-4 bytes per packet when the MID header extension is used and by 4-8 bytes when both header extensions are used. Bug: webrtc:10078 Change-Id: I5fa3ce28a75224adf11d2792bf4ff8dc76e46d99 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146480 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28685}
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.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct