Description
I used the Dockerfile to build a docker image and deployed an instance. While everything works fine, the webrtc previews in the web interface fails with the error "ERROR: Failed to connect WebRTC with server". The browser console states "New ICE connection state: failed".
With LOG_LEVEL=DEBUG, I get the below logs in the console.
DEBUG: [ output1] Creating with pipeline: intervideosrc name=intervideosrc timeout=86400000000000 ! videoconvert ! videoscale ! videorate ! capsfilter name=capsfilter ! vp8enc deadline=1 keyframe-max-dist=30 ! rtpvp8pay ! application/x-rtp,format=RGB,media=video,encoding-name=VP8,payload=97,width=480,height=270 ! tee name=webrtc_video_tee webrtc_video_tee. ! fakesink interaudiosrc name=interaudiosrc ! audioconvert ! level message=true ! audioresample name=webrtc-audioresample ! opusenc bandwidth=superwideband ! rtpopuspay ! application/x-rtp,media=audio,encoding-name=OPUS,payload=96 ! tee name=webrtc_audio_tee webrtc_audio_tee. ! fakesink
DEBUG: [ input1] Pipeline state change from PAUSED to PAUSED (pending PAUSED)
DEBUG: [ input1] Message from GStreamer: Latency from output1_interaudiosink_857589
DEBUG: [ output1] Move to state READY complete
INFO: [api_routes] Created output #1 with details {'type': 'webrtc', 'source': 'input1'}
DEBUG: [ output1] Pipeline state change from NULL to READY
DEBUG: [ output1] Move to state PAUSED has completed but no data yet
DEBUG: [ input1] Pipeline state change from PAUSED to PAUSED (pending PLAYING)
DEBUG: [ input1] Pipeline state change from PAUSED to PLAYING
DEBUG: [ output1] Pipeline state change from READY to PAUSED
DEBUG: [ output1] Move to state PLAYING is IN PROGRESS
DEBUG: [ output1] Message from GStreamer: Latency from opusenc0
DEBUG: [ output1] Message from GStreamer: Latency from vp8enc0
DEBUG: [ output1] Pipeline state change from PAUSED to PLAYING
INFO: [ output1] I now have 1 peers
DEBUG: [ output1] Property notify: object="webrtcbin0", property_name="stun-server", property_value="stun://stun.l.google.com:19302"
DEBUG: [ output1] Pipeline state change from PLAYING to PAUSED (pending READY)
DEBUG: [ output1] Pipeline state change from PAUSED to READY
DEBUG: [ output1] Pipeline state change from READY to PAUSED (pending PLAYING)
DEBUG: [ output1] Successfully added a new peer request
DEBUG: [ output1] Message from GStreamer: Latency from opusenc0
DEBUG: [ output1] Message from GStreamer: Latency from vp8enc0
DEBUG: [ output1] Pipeline state change from PAUSED to PLAYING
DEBUG: [ output1] Sending SDP offer to client (1204 chars in length)
DEBUG: [ output1] Property notify: object="webrtcbin0", property_name="signaling-state", property_value="enum GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER of type GstWebRTC.WebRTCSignalingState"
DEBUG: [ output1] Property notify: object="webrtcbin0", property_name="ice-gathering-state", property_value="enum GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE of type GstWebRTC.WebRTCICEGatheringState"
DEBUG: [ output1] Property notify: object="webrtcbin0", property_name="signaling-state", property_value="enum GST_WEBRTC_SIGNALING_STATE_STABLE of type GstWebRTC.WebRTCSignalingState"
DEBUG: [ output1] Property notify: object="webrtcbin0", property_name="ice-connection-state", property_value="enum GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING of type GstWebRTC.WebRTCICEConnectionState"
DEBUG: [ output1] Property notify: object="webrtcbin0", property_name="connection-state", property_value="enum GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING of type GstWebRTC.WebRTCPeerConnectionState"
DEBUG: [ output1] Property notify: object="webrtcbin0", property_name="ice-connection-state", property_value="enum GST_WEBRTC_ICE_CONNECTION_STATE_FAILED of type GstWebRTC.WebRTCICEConnectionState"
DEBUG: [ output1] Property notify: object="webrtcbin0", property_name="connection-state", property_value="enum GST_WEBRTC_PEER_CONNECTION_STATE_FAILED of type GstWebRTC.WebRTCPeerConnectionState"
DEBUG: [ output1] In PLAYING state: pipeline, queue9, queue8, webrtcbin0, capsfilter8, capsfilter7, fakesink3, webrtc_audio_tee, rtpopuspay0, opusenc0, webrtc-audioresample, level0, audioconvert1, interaudiosrc, fakesink2, webrtc_video_tee, rtpvp8pay0, vp8enc0, capsfilter, videorate0, videoscale2, videoconvert2, intervideosrc
Enabling more verbose logging spits out too much gibberish that I can't make sense of. How do I make the webrtc previews work?