| commit | f475d365a25036725c3f545f57de59d2cc902d17 | [log] [tgz] |
|---|---|---|
| author | Taylor Brandstetter <deadbeef@webrtc.org> | Fri Jan 08 23:35:57 2016 |
| committer | Taylor Brandstetter <deadbeef@webrtc.org> | Fri Jan 08 23:36:06 2016 |
| tree | d64f2ab48eba9691faa53b57e77528ea5048bfd5 | |
| parent | 25702cb1628941427fa55e528f53483f239ae011 [diff] |
Properly handle different transports having different SSL roles. This meant splitting "transport_options" into audio/video/data options, for when creating the answer, and giving "GetSslRole" a "transport_name" parameter so we can retrieve the current role on a per-transport basis. BUG=webrtc:4525 R=pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/1516993002 . Cr-Commit-Position: refs/heads/master@{#11192}
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.