93e177862c
Directly use RtpPacketToSend instead of RtpPacketSendInfo For RFC8888 we will need to match SSRC and RTP sequence number with the transport sequence number that only exist on the sending side. For retransmitted packets, RTX, RtpPacketSendInfo contain original SSRC and original sequence number. Bug: webrtc:42225697 Change-Id: Iafa5d851ca5c51c85e4607ed4c1919d96da6084a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366000 Auto-Submit: Per Kjellander <perkj@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43264}
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