f72c06419f
Prior to this commit, the SCTP association could terminate due to too many retransmission attempts when there is a long duration of packet loss. The RTCPeerConnection wouldn't terminate, and when the network later recovers (possibly using a different ICE candidate), it would be a RTCPeerConnection with media, but without DataChannels. This commit will make the dcSCTP library never abort by itself when there are too many retransmissions. It will also put a cap on the retry duration so that it will do a retry every three seconds (or lower). Bug: webrtc:13129 Change-Id: I08162ea20d6a60aa0eae2717966d9a2ddba8fc22 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232540 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35061}
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