8267724a85
The receive buffer mustn't be full; If it's full, and a message can't be assembled, the socket can't accept more data. To avoid this, there is a high watermark limit that, when reached, will make the socket only accept chunks that advance the cumulative ack TSN. Before this CL, the announced receiver window size in every sent SACK was based on what the receive buffer could maximally be, which means that in really high data rate applications, the amount of outstanding data could actually fill the receive buffer (due to packet loss, that prevents messages from being reassembled). As the socket started behaving more conservatively when the high watermark limit was reached, this resulted in unnecessary T3-RTXes. But by announcing the high watermark limit instead, the sender will stay within it, and will have a peer socket that behaves as expected. Bug: webrtc:12799 Change-Id: Ife2f409914a230640217553c54f60d05843efc70 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220762 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34168}
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