| commit | 824f58621315a8b39193e06f960d8ff15f84fc13 | [log] [tgz] |
|---|---|---|
| author | deadbeef <deadbeef@webrtc.org> | Wed Aug 24 22:06:53 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Wed Aug 24 22:06:58 2016 |
| tree | fc498efa9175d0bf75f9d47247cd223154ee7beb | |
| parent | 1d7a637340ea097ea55061af7b566ceccc2678c8 [diff] |
Fixing segfault caused by TurnServer. TURN server sockets were being destroyed asynchronously, which could happen after the TurnServer itself (and even the VirtualSocketServer used by the sockets) were destroyed. This is fixed easily by using an AsyncInvoker (to ensure the async operation doesn't occur after its initiator is destroyed), and keeping the objects waiting for deletion in a unique_ptr vector. Review-Url: https://codereview.webrtc.org/2264343002 Cr-Commit-Position: refs/heads/master@{#13907}
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.