webrtcbin: Is there a way to identify that WebRTC connection is closed?
From documentation at least there doesn't seem to be a clear way to catch an event of a remote client closing/dropping WebRTC connection. Is it true? If so, then it is hard to imagine WebRTC implementation being usable in practice.
UPD: Sad part if that it causes problems that shouldn't exist in Rust.
With pipeline that has
webrtcbin -> decodebin -> autoaudiosink for audio part it fails like this:
0:00:50.850212965 16200 0x7f6874001c70 WARN pulse pulsesink.c:702:gst_pulsering_stream_underflow_cb:<autoaudiosink0-actual-sink-pulse> Got underflow fish: “env GST_DEBUG=3 cargo run” terminated by signal SIGSEGV (Address boundary error) Exit code 139