ed447aedb8
This commit makes the following changes: 1. Splits TestReconstructedServerUrl into 3 tests that individually check the reconstructed URL for UDP IPv4, UDP IPv6, and TCP. 2. Factors out common code between protocols for release allocation and reconstructed URL tests. 3. Provides functions for getting the expected RTT of various operations based on the protocol used. TurnPort tests use a fake clock in part to check tight bounds on the number of network round trips it takes to complete operations like getting TURN candidates and trying alternate servers. These RTTs are highly dependent on the characteristics of the transport-layer protocol used, so these details have been moved to dedicated functions with comments explaining how the numbers are calculated. Bug: webrtc:7584 Change-Id: I3b065e25446cb5ecd955f359625a35fb0ad46777 Reviewed-on: https://chromium-review.googlesource.com/611500 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19395}
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.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc