<html><head><meta http-equiv="Content-Type" content="text/html; charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""><div class="">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 '<span style="font-family: Menlo; font-size: 11px;" class="">WARNING: erroneous pipeline: syntax error</span>’ when I run the command. </div><div class="">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?</div><div class=""><br class=""></div><div class="">gst-launch-1.0 -e \</div><div class=""> uridecodebin uri="<a href="file:///home/fedora/starwars.mov" class="">file:///home/fedora/starwars.mov</a>" \</div><div class=""> ! qtdemux name=demux demux.audio_0 \</div><div class=""> ! queue \</div><div class=""> ! audioconvert \</div><div class=""> ! opusenc bandwidth=superwideband bitrate-type=vbr \</div><div class=""> ! rtpopuspay \</div><div class=""> ! rtprtxqueue max-size-time=2000 max-size-packets=0 \</div><div class=""> ! udpsink host=<a href="http://www.playbacktc.com" class="">www.playbacktc.com</a> port=5052 \</div><div class=""> demux.video_0 \</div><div class=""> ! queue \</div><div class=""> ! videorate ! video/x-raw, framerate=30000/1001 \</div><div class=""> ! videoconvert \</div><div class=""> ! x264enc tune=zerolatency speed-preset=1 dct8x8=true quantizer=17 pass=qual \</div><div class=""> ! rtph264pay \</div><div class=""> ! rtprtxqueue max-size-time=2000 max-size-packets=0 \</div><div class=""> ! rtpbin.send_rtp_sink_0 \</div><div class=""> ! udpsink host=<a href="http://www.playbacktc.com" class="">www.playbacktc.com</a> port=5054 \</div></body></html>