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When the transport is terminated, if an error has occured, it will be propagated to the channels. When such errors can happen at the SCTP level (e.g. out of resources), RTCError may contain an error code matching the definition at https://www.iana.org/assignments/sctp-parameters/sctp-parameters.xhtml#sctp-parameters-24 If the m= line is rejected or removed from SDP, an error will again be sent to the data channels, signaling their unexpected transition to closed. Bug: webrtc:12904 Change-Id: Iea3d8aba0a57bbedb5d03f0fb6f7aba292e92fe8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223541 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34386}
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