Sebastian Jansson e6cedbbff6 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}
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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 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.

More info

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