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The retransmission timeout (RTO) value is updated on every measured RTT and is a function of the RTT value and its stability. In reality, the RTT is never constant - it fluctuates, which makes the RTO become much larger than the RTT. But for extremely stable RTTs, which we get in simulations, the RTO value can become the same as the RTT, and that makes expiration timers be scheduled to the RTT value, and will race with packets that are expected to stop the expiration timer. And that race should be avoided in simulations. So ensuring that the RTO value is always greater, if only be a single millisecond, will work fine in these simulations. Bug: webrtc:12614 Change-Id: I30cf9c97e50449849ab35de52696c618d8498128 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219680 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34084}
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 here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs