02af84064c
With this cl RtpTransportControllerSend::OnAddPacket is instead directly invoked from PacketRouter::SendPacket instead of going via RTP module. Transport sequence numbers are instead of directly written to header extension, added to RtpPacketToSendMetaData and written to the extenion by RTP module. This is to allow transport sequence numbers without actually sending them in an extension. Bug: webrtc:15368 Change-Id: Idd03e02a4257dfc4d0f1898b2803345975d7dad2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344720 Reviewed-by: Erik Språng <sprang@google.com> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41974}
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 here 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: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation