8df32eb0e1
Before this change, there was no way for a client to indicate to the dcSCTP library if a packet that was supposed to be sent, was actually sent. It was assumed that it always was. To handle temporary failures better, such as retrying to send packets that failed to be sent when the send buffer was full, this information is propagated to the library. Note that this change only covers the API and adaptations to clients. The actual implementation to make use of this information is done as a follow-up change. Bug: webrtc:12943 Change-Id: I8f9c62e17f1de1566fa6b0f13a57a3db9f4e7684 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228563 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34767}
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