| commit | 14f97f5bc65f2319642719d5a061a3747f586bb7 | [log] [tgz] |
|---|---|---|
| author | deadbeef <deadbeef@webrtc.org> | Thu Jun 23 00:14:15 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Thu Jun 23 00:14:20 2016 |
| tree | 3ec428094115e37ab4e42993139eac5df11aad74 | |
| parent | e7edea97592733649ac067355da122b047b128cd [diff] |
Adding IceConfig option to assume TURN/TURN candidate pairs will work. This will allow media to be sent on these pairs before a binding response is received, shortening call setup time. However, this is only possible if the TURN servers don't require CreatePermission when communicating with each other. R=honghaiz@webrtc.org, pthatcher@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2063823008 Cr-Commit-Position: refs/heads/master@{#13268}
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.