| commit | c67e0f5753e9055e5fe76ae5eba76fe9efb3d769 | [log] [tgz] |
|---|---|---|
| author | Honghai Zhang <honghaiz@webrtc.org> | Mon Sep 19 23:57:37 2016 |
| committer | Honghai Zhang <honghaiz@webrtc.org> | Mon Sep 19 23:57:48 2016 |
| tree | 9e35b6f754cf73b184d07d153b3d1393db617792 | |
| parent | 042041bf9585f92e962387c59ca805f1218338f9 [diff] |
Signal to remove remote candidates if ports are pruned. Previously when a Turn port is pruned, if its candidate has been sent to the remote side, the remote side will keep the candidate and use that to create connections. We now signal the remote side to remove the candidates so that at least no new connection will be created using the removed candidates. Also updated the virtual socket server to better support our test cases. 1. Allow the virtual socket server to set transit delay for packets sent from a given IP address. 2. Ensure the ordered packet delivery for each socket (Previously the delivery order is enforced on the whole test case, so if a udp packet gets delayed based on its IP address, all TCP packets sent after the UDP packet will be delayed at least until the UDP packet is received). BUG=webrtc:6380 R=deadbeef@webrtc.org, pthatcher@webrtc.org, skvlad@webrtc.org Review URL: https://codereview.webrtc.org/2261523004 . Cr-Commit-Position: refs/heads/master@{#14297}
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.