gstreamer/subprojects/gst-plugins-good/tests/examples/rtp/client-H263p-AMR.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)H263-1998"
AUDIO_CAPS="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)AMR,encoding-params=(string)1,octet-align=(string)1"
VIDEO_DEC="rtph263pdepay ! avdec_h263"
AUDIO_DEC="rtpamrdepay ! amrnbdec"
VIDEO_SINK="videoconvert ! autovideosink"
AUDIO_SINK="audioconvert ! audioresample ! autoaudiosink"
gst-launch-1.0 -v rtpbin name=rtpbin latency=100 \
udpsrc caps=$VIDEO_CAPS port=5000 ! rtpbin.recv_rtp_sink_0 \
rtpbin. ! $VIDEO_DEC ! $VIDEO_SINK \
udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false \
udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_1 \
rtpbin. ! $AUDIO_DEC ! $AUDIO_SINK \
udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \
rtpbin.send_rtcp_src_1 ! udpsink port=5007 sync=false async=false