ba90b7f17184db8f0a1683593b5af53631e9b954

The a=rtcp:9 IN IP4 0.0.0.0 line is required by JSEP to be generated, but is also required to be ignored. This reduces log spew. Bug: None Change-Id: I984060d9693b9df4c4cfdf2c5dea0ea620f4bc83 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196641 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Philipp Hancke <philipp.hancke@googlemail.com> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32798}
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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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct
- Reporting bugs
Description
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