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gst-plugins-rtp

This is a re-implementation of the RTP elements that are submitted in 2013 to handle RTP streams. The elements handle a correct connection for the bi-directional use of the RTCP sockets.

https://bugzilla.gnome.org/show_bug.cgi?id=703111

The nrtp_rtpsink and nrtp_rtpsrc elements add an URI interface so that streams can be decoded with decodebin using the rtp:// interface.

NOTE: these modules will be part of a GStreamer bad plugins >= 1.18.0.

The code can be used as follows

gst-launch-1.0 videotestsrc ! x264enc ! rtph264pay config-interval=3 ! nrtp_rtpsink uri=rtp://239.1.1.1:1234
gst-play-1.0 rtp://239.1.1.1:1234

gst-launch-1.0 videotestsrc ! x264enc ! rtph264pay config-interval=1 ! nrtp_rtpsink uri=rtp://239.1.2.3:5000
gst-launch-1.0 nrtp_rtpsrc uri=rtp://239.1.2.3:5000?encoding-name=H264 ! rtph264depay ! avdec_h264 ! videoconvert ! xvimagesink

gst-launch-1.0 videotestsrc ! avenc_mpeg4 ! rtpmp4vpay config-interval=1 ! nrtp_rtpsink uri=rtp://239.1.2.3:5000
gst-launch-1.0 nrtp_rtpsrc uri=rtp://239.1.2.3:5000?encoding-name=MP4V-ES ! rtpmp4vdepay ! avdec_mpeg4 ! videoconvert ! xvimagesink

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GStreamer Element to handle RTP/RTCP

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