WebRTC example doesn't show remote test stream
Describe your issue
I'm trying out t he webrtc example and I can't seem to get the remote stream to show.
I'm running the signalling server:
WEBRTCSINK_SIGNALLING_SERVER_LOG=debug cargo run --bin gst-webrtc-signalling-server
...
2023-11-13T02:44:55.110603Z INFO ThreadId(01) gst_webrtc_signalling_server: Listening on: 0.0.0.0:8443
2023-11-13T02:45:10.965463Z INFO ThreadId(01) gst_webrtc_signalling_server: Accepting connection from 127.0.0.1:57614
2023-11-13T02:45:10.965536Z DEBUG ThreadId(32) accept_async: gst_plugin_webrtc_signalling::server: new
2023-11-13T02:45:10.967679Z DEBUG ThreadId(04) accept_async: log: Server handshake done.
2023-11-13T02:45:10.967697Z INFO ThreadId(04) accept_async: gst_plugin_webrtc_signalling::server: New WebSocket connection this_id=30ce7f3a-2e4d-4658-ab52-2553fbd4f543
2023-11-13T02:45:10.967712Z DEBUG ThreadId(04) accept_async: gst_plugin_webrtc_signalling::server: close time.busy=110µs time.idle=2.08ms
2023-11-13T02:45:10.997742Z INFO ThreadId(31) gst_plugin_webrtc_signalling::server: Received message Ok(Text("{\"type\":\"setPeerStatus\",\"roles\":[\"listener\"],\"meta\":{\"name\":\"WebClient-1699843486430\"}}"))
2023-11-13T02:45:10.997804Z DEBUG ThreadId(31) set_peer_status{peer_id="30ce7f3a-2e4d-4658-ab52-2553fbd4f543" status=PeerStatus { roles: [Listener], meta: Some(Object {"name": String("WebClient-1699843486430")}), peer_id: None }}: gst_plugin_webrtc_signalling::handlers: new
2023-11-13T02:45:10.997824Z INFO ThreadId(31) set_peer_status{peer_id="30ce7f3a-2e4d-4658-ab52-2553fbd4f543" status=PeerStatus { roles: [Listener], meta: Some(Object {"name": String("WebClient-1699843486430")}), peer_id: None }}: gst_plugin_webrtc_signalling::handlers: registered as a producer peer_id=30ce7f3a-2e4d-4658-ab52-2553fbd4f543
2023-11-13T02:45:10.997834Z DEBUG ThreadId(31) set_peer_status{peer_id="30ce7f3a-2e4d-4658-ab52-2553fbd4f543" status=PeerStatus { roles: [Listener], meta: Some(Object {"name": String("WebClient-1699843486430")}), peer_id: None }}: gst_plugin_webrtc_signalling::handlers: close time.busy=17.2µs time.idle=13.4µs
The frontend:
npm start
...
<i> [webpack-dev-server] [HPM] Upgrading to WebSocket
<i> [webpack-dev-server] [HPM] Upgrading to WebSocket
<i> [webpack-dev-server] [HPM] Client disconnected
<i> [webpack-dev-server] [HPM] Client disconnected
<i> [webpack-dev-server] [HPM] Client disconnected
<i> [webpack-dev-server] [HPM] Upgrading to WebSocket
The test stream:
gst-launch-1.0 webrtcsink name=ws meta="meta,name=gst-stream" videotestsrc ! ws. audiotestsrc ! ws.
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
Redistribute latency...
Redistribute latency...
Got context from element 'vaapiencodeh264-0': gst.vaapi.Display=context, gst.vaapi.Display=(GstVaapiDisplay)"\(GstVaapiDisplayGLX\)\ vaapidisplayglx0", gst.vaapi.Display.GObject=(GstObject)"\(GstVaapiDisplayGLX\)\ vaapidisplayglx0";
^Chandling interrupt.
Interrupt: Stopping pipeline ...
Execution ended after 0:00:18.734306492
Setting pipeline to NULL ...
^C
Expected Behavior
I would assume the live feed should show up?
Observed Behavior
I don't see it.
Setup
- Operating System: Linux Mint (based on Ubuntu 22.04)
- Device: Computer
- gst-plugins-rs Version: 6c5c09fa
- GStreamer Version: GStreamer 1.20.3
- Command line:
Steps to reproduce the bug
- open terminal
- type
command
How reproducible is the bug?
Always for me.
Screenshots if relevant
Solutions you have tried
Reloading endlessly, staring blankly.
Related non-duplicate issues
No idea.
Additional Information
I'm a WebRTC beginner, I'm probably just doing it wrong...