commit | 1d8a506405734d0cef9653704b036ca4f1388960 | [log] [tgz] |
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author | stefan <stefan@webrtc.org> | Fri Oct 02 10:39:33 2015 |
committer | Commit bot <commit-bot@chromium.org> | Fri Oct 02 10:39:40 2015 |
tree | 610cbeb60219151c252e7f4fa693c590900ee452 | |
parent | da903eaabbb6c6830efcafc3c2ade1d36f511e43 [diff] |
Add a PacketOptions struct to webrtc::Transport. This allows us to pass packet meta data, such as transport sequence number, to libjingle and further down to the socket implementation. A similar struct already exist in libjingle, see rtc::PacketOptions in asyncpacketsocket.h. BUG=4173 Review URL: https://codereview.webrtc.org/1376673004 Cr-Commit-Position: refs/heads/master@{#10144}
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.