commit | 398c3fd6c220531a05315dfe3ce871ec64686402 | [log] [tgz] |
---|---|---|
author | zstein <zstein@webrtc.org> | Wed Jul 19 20:38:02 2017 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Jul 19 20:38:02 2017 |
tree | 1da89eba05e397d9dc780b57dc773dcecb1463ba | |
parent | f6a861ab6c767cfb9579f43cba7f5e55a09e3848 [diff] |
Introduce RtpTransportInternal and SrtpTransport. SrtpTransport currently just delegates everything to RtpTransport. Also makes BaseChannel::rtp_transport_ an RtpTransportInternal and constructs an SrtpTransport if srtp is required. BUG=webrtc:7013 Review-Url: https://codereview.webrtc.org/2981013002 Cr-Commit-Position: refs/heads/master@{#19095}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.