Minor Dtls-in-STUN-encapsulation cleanups That unfortunately does not solve the existing bug :( 1) Add a new method GetPending that is used when retransmitting unencapsulated, this only matters for PQC and means that we might send 2 packets "back-to-back", which is how it's ordinarely done. 2) Add a new UpdateHandshakeTimeout, that is used after the initial estimate and pipe that all the way to rtc_base/openssl_stream_adapter.cc. 3) Move the RunBenchmark into the test fixture, that way it can easily be used from other tests. 4) Modify the "with packetloss" test to run with 25% packetloss, from 50%. This might seem like a bummer, but at 50% the tests very super fragile, changing *anything* made them randomly fail. I.e. we had overfitted the settings *a lot*. At 25% they are stable. 5) Add a new test that verifies that using dtls-in-stun leads to short connect time that not using it. BUG=webrtc:367395350 Change-Id: I9cc207eab413dc0e7542464174e504a39f425466 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/435182 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46524}
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.