commit | e6cedbbff68534e846385a8190a3b0d39fe60717 | [log] [tgz] |
---|---|---|
author | Sebastian Jansson <srte@webrtc.org> | Mon Mar 09 19:25:18 2020 +0100 |
committer | Commit Bot <commit-bot@chromium.org> | Thu Mar 12 17:42:13 2020 +0000 |
tree | 99fb2c9516750ea9811e28a92f38932fc09aa726 | |
parent | d35a6865178acf1596ae6674dad894b47c4f0a23 [diff] |
Ensures that all simulated TCP packets are at least 4 bytes. Bug: webrtc:10839 Change-Id: I4f2f5cf75b9fbcedb39e3fa05d11c68a7de6f5b1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170051 Reviewed-by: Ali Tofigh <alito@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30780}
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 http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.