I am running the Janus Video Room demo in the webrtc examples and am a noob when it comes to intricacies of Gstreamer. I am encountering the following errors when connecting to my janus room. All Janus communication appear normal.
(<unknown>:64485): GStreamer-CRITICAL **: 11:59:51.845: gst_bin_add: assertion 'GST_IS_ELEMENT (element)' failed
(<unknown>:64485): GStreamer-CRITICAL **: 11:59:51.845: gst_element_sync_state_with_parent: assertion 'GST_IS_ELEMENT (element)' failed
(<unknown>:64485): GStreamer-CRITICAL **: 11:59:51.846: gst_element_link_pads_full: assertion 'GST_IS_ELEMENT (src)' failed
(<unknown>:64485): GStreamer-CRITICAL **: 11:59:51.847: Element rtpbin already has a pad named send_rtp_sink_0, the behaviour of gst_element_get_request_pad() for existing pads is undefined!
** (<unknown>:64485): WARNING **: 11:59:51.847: (gstwebrtcbin.c:2917):_connect_input_stream: code should not be reached
Is it safe to assume this is an error generated after I start the pipeline?
def start_pipeline(self):
self.pipe = Gst.parse_launch(PIPELINE_DESC)
self.webrtc = self.pipe.get_by_name(‘sendrecv')
self.webrtc.connect('on-negotiation-needed', self.on_negotiation_needed)
self.webrtc.connect('on-ice-candidate', self.send_ice_candidate_message)
self.webrtc.connect('pad-added', self.on_incoming_stream)
trans = self.webrtc.emit('get-transceiver', 0)
if DO_RTX:
trans.set_property ('do-nack', True)
self.pipe.set_state(Gst.State.PLAYING)
Tyler
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