| commit | afee708f66ba0581f29446166a56c691919ea19e | [log] [tgz] |
|---|---|---|
| author | Philipp Hancke <philipp.hancke@googlemail.com> | Thu Oct 22 09:55:58 2020 |
| committer | Commit Bot <commit-bot@chromium.org> | Fri Oct 23 20:14:53 2020 |
| tree | f1bb1ccd0514323fd10cfd11615400dcd6d1f137 | |
| parent | 3d25935127400418ef6b5970f0edec5db01d2ba6 [diff] |
do not set rtp datachannel b=AS for SCTP the limit is ignored anyway. Also rename rtp datachannel bandwidth limit constant. BUG=webrtc:6625 Change-Id: If7b26691ced8148955e98c86b9bed692b2e55e8e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189972 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32479}
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.
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.