9ff75a6206
Before: ``` (stun_port.cc:596): sendto : [0x00000041] No route to host ``` After: ``` (stun_port.cc:598): UDP send of 20 bytes to host stun1.l.google.com:19302 (74.125.200.127:19302) failed with error 65 : [0x00000041] No route to host ``` Bug: None Change-Id: Ibcd487e97b37677225814562df30af66f655cddb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215000 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com> Cr-Commit-Position: refs/heads/master@{#33694}
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