4c990e2e56
The limits have been moved out from the Send Queue as they were enforced outside the queue anyway (in the socket). That was a preparation for adding even more limits; There is now also a per-stream limit, allowing individual streams to have one (global) limit, and the entire socket to have another limit. These limits are very small in the default options. In Chrome, the limit is 16MB per stream, so expect the defaults to be updated when the additional buffering outside dcSCTP is removed. Bug: chromium:41221056 Change-Id: I9f835be05d349cbfce3e9235d34b5ea0e2fe87d1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342481 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41895}
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