| commit | 6bb1ef2b868fef645fdddb41599d6a6693b634d8 | [log] [tgz] |
|---|---|---|
| author | Taylor Brandstetter <deadbeef@webrtc.org> | Tue Jun 28 01:09:03 2016 |
| committer | Taylor Brandstetter <deadbeef@webrtc.org> | Tue Jun 28 01:09:10 2016 |
| tree | 7aebbe1c703796c610472929a76e31a86163c2e0 | |
| parent | f8e65779a7dc890d8d6f38eca50985a2f0407da1 [diff] |
Fixing bug where Connection drops packets when presumed writable. The "should I simulate EWOULDBLOCK?" determination now happens solely in P2PTransportChannel. This also fixes a bug where the "last packet id" was set even if no packet was sent. R=honghaiz@webrtc.org, pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/2099783002 . Cr-Commit-Position: refs/heads/master@{#13307}
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. This page is maintained by the Google Chrome team.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.