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The socket can measure the round-trip-time (RTT) by two different scenarios: * When a sent data is ACKed * When a HEARTBEAT has been sent, which as been ACKed. The RTT will be used to calculate which timeout value that should be used for e.g. the retransmission timer (T3-RTX). On connections with a low RTT, the RTO value will be low, and on a connection with high RTT, the RTO value will be high. And on a connection with a generally low RTT value, but where it varies a lot, the RTO value will be calculated to be fairly high, to not fire unnecessarily. So jitter is bad, and is part of the calculation. Bug: webrtc:12614 Change-Id: I64905ad566d5032d0428cd84143a9397355bbe9f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214045 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33832}
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