commit | 72a43a1d2cf3a667bbd218383aa975de72ed6804 | [log] [tgz] |
---|---|---|
author | Qingsi Wang <qingsi@google.com> | Wed Feb 21 00:03:18 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Feb 21 00:49:26 2018 |
tree | 4ca655d4d75c9611eca8e302635bc80b41b8c09b | |
parent | 54b8407ee59360b1744a8db5958437e7ca566d9d [diff] |
Collect packet loss and RTT stats of STUN binding requests. STUN candidates use STUN binding requests to keep NAT bindings open. Related stats including packet loss and RTT can be now collected via the legacy GetStats in PeerConnection. Bug: None Change-Id: I7b0eee1ccb07eb670a32ee303c9590047b25f31c Reviewed-on: https://webrtc-review.googlesource.com/54100 Commit-Queue: Qingsi Wang <qingsi@google.com> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22113}
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.