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Some deployments, e.g. Chromium, has a limited send buffer. It's reasonable that it's quite small, as it avoids queuing too much, which typically results in increased latency for real-time communication. To avoid SCTP to fill up the entire buffer at once - especially when doing fast retransmissions - limit the amount of packets that are sent in one go. In a typical scenario, SCTP will not send more than three packets for each incoming packet, which is is the case when a SACK is received which has acknowledged two large packets, and which also adds the MTU to the congestion window (due to in slow-start mode), which then may result in sending three packets. So setting this value to four makes any retransmission not use that much more of the send buffer. This is analogous to usrsctp_sysctl_set_sctp_fr_max_burst_default in usrsctp, which also has the default value of four (4). Bug: webrtc:12943 Change-Id: Iff76a1668beadc8776fab10312ef9ee26f24e442 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228480 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34744}
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