Use the matching relay port when using TURN servers as STUN servers PortConfiguration::GetRelayServerAddresses() checked whether a relay config had a port with the requested protocol, but then returned the address of the config's first port. For a config listing e.g. a TCP port before a UDP one, STUN binding requests were sent over UDP to the TCP address, so no srflx candidate was gathered from that server and gathering completion was delayed until the requests timed out. Return the addresses of all ports that match the protocol instead. RelayServerConfigs created from RTCIceServer URLs have a single port, so only configs built directly through the native API are affected. Bug: webrtc:573252240 Change-Id: I260c0f0afc3ddf61b3f34163e2b3a5a257b7575d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/508721 Commit-Queue: Kacper Wasniowski <kwasniow@cisco.com> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#48911}
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.
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.