023be3c977
Create a server using: ./data_channel_benchmark --server --port 12345 Start the flow of data from the server to a client using: ./data_channel_benchmark --port 12345 --transfer_size 100 The throughput is reported on the server console. The negotiation does not require a 3rd party server and is done over a gRPC transport. No TURN server is configured, so both peers need to be reachable using STUN only. Bug: webrtc:13288 Change-Id: Iac9a96cf390ab465ea45a46bf0b40950c56dfceb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235661 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36206}
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