Experiments with full duplex audio streaming in gstreamer Expected behaviour: full duplex stream Observed behaviour: does not start sending until data is received. AUDIO_CAPS="audio/x-raw,format=S16LE,rate=24000" RTP_CAPS="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS,payload=(int)96" REMOTE=192.168.1.10 gst-launch-1.0 -v rtpbin name=rtpbin latency=100 \ audiotestsrc is-live=true ! $AUDIO_CAPS ! opusenc ! rtpopuspay ! rtpbin.send_rtp_sink_0 \ rtpbin.send_rtp_src_0 ! udpsink host=$REMOTE port=5000 \ rtpbin.send_rtcp_src_0 ! udpsink sync=false async=false host=$REMOTE port=5001 \ udpsrc port=5000 caps=$RTP_CAPS ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtpopusdepay ! opusdec ! autoaudiosink \ udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0