| commit | 14461d42bc7ad3710380083cf8857baf61e8952a | [log] [tgz] |
|---|---|---|
| author | deadbeef <deadbeef@webrtc.org> | Wed Jun 15 18:06:57 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Wed Jun 15 18:07:05 2016 |
| tree | 6cd4b4cacf8a598f8ea5e33ae1486863278eb3e0 | |
| parent | a6219cc3ef08dd9b2981b065b6f051d7de0866f8 [diff] |
Fixing flaky test: WebRtcSessionTest.TestPacketOptionsAndOnPacketSent The test sent a media packet, then verified it was sent by checking the "last packet sent"'s ID. But the last packet sent may have been a STUN packet that came *after* the media packet. BUG=webrtc:5978 Review-Url: https://codereview.webrtc.org/2071573002 Cr-Commit-Position: refs/heads/master@{#13156}
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.