d68f18ee6e
This is mentioned in https://datatracker.ietf.org/doc/html/rfc4960#section-6.3.1 and further described in https://datatracker.ietf.org/doc/html/rfc6298#section-4. The TCP RFCs mentioned G as the clock granularity, but in SCTP it should be set much higher, to account for the delayed ack timeout (ATO) of the peer (as that can be seen as a very high clock granularity). That one is set to 200ms by default in many clients, so a reasonable default limit could be set to 220 ms. If the measured variance is higher, it will be used instead. Bug: webrtc:12943 Change-Id: Ifc217daa390850520da8b3beb0ef214181ff8c4e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232614 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35068}
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
- Documentation