gstreamer/subprojects/gst-plugins-good/tests/examples/rtp/client-VP8-OPUS.sh

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#!/bin/sh
#
# A simple RTP receiver
#
VIDEO_CAPS="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)VP8"
AUDIO_CAPS="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS"
SRC=localhost
DEST=localhost
VIDEO_DEC="rtpvp8depay ! vp8dec"
AUDIO_DEC="rtpopusdepay ! opusdec"
VIDEO_SINK="videoconvert ! autovideosink"
AUDIO_SINK="audioconvert ! audioresample ! autoaudiosink"
LATENCY=100
gst-launch-1.0 -v rtpbin name=rtpbin latency=$LATENCY \
udpsrc caps=$VIDEO_CAPS address=$SRC port=5000 ! rtpbin.recv_rtp_sink_0 \
rtpbin. ! $VIDEO_DEC ! $VIDEO_SINK \
udpsrc address=$SRC port=5001 ! rtpbin.recv_rtcp_sink_0 \
rtpbin.send_rtcp_src_0 ! udpsink host=$DEST port=5005 sync=false async=false \
udpsrc caps=$AUDIO_CAPS address=$SRC port=5002 ! rtpbin.recv_rtp_sink_1 \
rtpbin. ! $AUDIO_DEC ! $AUDIO_SINK \
udpsrc address=$SRC port=5003 ! rtpbin.recv_rtcp_sink_1 \
rtpbin.send_rtcp_src_1 ! udpsink host=$DEST port=5007 sync=false async=false