Streaming audio and video RTP

[Patrick Cusack]

I have a endpoint that expects audio and video over ports 5052 and 5054 respectively. I am using the following script to send audio and video. I am getting a 'WARNING: erroneous pipeline: syntax error’ when I run the command. 
Also, does using simple rtp payloads into a udp sink bypass RTCP feedback, ie if my server is NACKing on account of dropped packets, does this hinder retransmission of rtp packets?

gst-launch-1.0 -e \
        uridecodebin uri="file:///home/fedora/starwars.mov" \
        ! qtdemux name=demux  demux.audio_0 \
        ! queue \
        ! audioconvert \
        ! opusenc bandwidth=superwideband bitrate-type=vbr \
        ! rtpopuspay  \
        ! rtprtxqueue max-size-time=2000 max-size-packets=0 \
        ! udpsink host=www.playbacktc.com port=5052 \
        demux.video_0 \
        ! queue \
        ! videorate ! video/x-raw, framerate=30000/1001 \
        ! videoconvert \
        ! x264enc tune=zerolatency speed-preset=1 dct8x8=true quantizer=17 pass=qual \
        ! rtph264pay \
        ! rtprtxqueue max-size-time=2000 max-size-packets=0 \
        ! rtpbin.send_rtp_sink_0 \
        ! udpsink host=www.playbacktc.com port=5054 \
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