1b83a9e400
Previously, each RTCP packet was handled several times in a row, once per m-section. This caused various weirdness and log warning spam, in particular when using unified plan. The cause was that the packets were wired trough each BaseChannel instance up to the Call class. With this fix, the RTCP packets are wired once per RtpTransportInternal via the common peer connection class. Bug: chromium:1002875 Change-Id: I41c4eb3b68e215ebe0f2c6fb93ae0ee73335b89a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152668 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29226}
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