Here is what I\'m trying:
gst-launch -v udpsrc port=1234 ! fakesink dump=1
I test with:
gst-launch -v audiotestsrc ! udpsink host=
Maybe packets are too large for udp? They are limited to 64K. Try resizing frames to really small size to check if this is the reason. If so, you may be interested in some compression and payloaders/depayloaders (gst-inspect | grep pay
).
gstreamer1-1.16.0-1.fc30
gst-launch-1.0 -v filesrc location=/.../.../.../sample-mp4-file.mp4 ! qtdemux ! h264parse ! queue ! rtph264pay config-interval=10 pt=96 ! udpsink port=8888 host=127.0.0.1
https://en.wikipedia.org/wiki/RTP_audio_video_profile
Try these (You may have to install gstreamer-ugly plugins for this one)
UDP streaming from Webcam (stream over the network)
gst-launch v4l2src device=/dev/video0 ! 'video/x-raw-yuv,width=640,height=480' ! x264enc pass=qual quantizer=20 tune=zerolatency ! rtph264pay ! udpsink host=127.0.0.1 port=1234
UDP Streaming received from webcam (receive over the network)
gst-launch udpsrc port=1234 ! "application/x-rtp, payload=127" ! rtph264depay ! ffdec_h264 ! xvimagesink sync=false
Update
To determine the payload at the streaming end simply use verbose option with gst-launch -v ...