e1e94ad4c8
With this cl, packets that are discarded in RtpTransport now notifies Call, so that they can be part of BWE even if they are dropped. These packets have been recevied on the transport, and has bin decrypted and parsed and thus can be accounted for. The un demuxable packets are forwarded to Call similarly how RTCP packets are forwarded. Bug: webrtc:14928 Change-Id: Ia53349c7b316c4442a3c7aac085a85ec4f4ab9ae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299262 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39727}
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