Due to a limit socket send buffer, it's quite easy to fill it up when using exponential slow start, which results in dropping a lot of packets and having to retransmit those. Disabling this, to align it to how SCTP normally behaves, and then try to stabilize it later. With SCTP slow start, it will increase with one MTU for each RTT when there is no packet loss. Even this mode will experience packet loss, but not as much will be lost, and it will stabilize quicker. Bug: webrtc:12614 Change-Id: Ibc484b19b7e708fe5bd837bbef178a2f69b7211f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218203 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33969}
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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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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