35f73ddca4
Modify openssl_stream_adapter to check/set timer regardless of dtls state. This is needed for DTLS1.3 orelse handshake will never complete if last client packet is lost (e.g if retransmit is not triggered after writable) as show by TestHandshakeLoseSecondClientPacket. TestHandshakeLoseSecondClientPacket works with/without this patch if using DTLS1.2. BUG=webrtc:383141571 Change-Id: I2757783c9e79686d1fbe0eff12341ab9e3863fdd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/372201 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43610}
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