Hi I'm currently trying to use gstreamer-1.0 to consume rtsp source and provide RTP streams for audio and video in the streams (for Janus Gateway). I have a working solution with ffmpeg so basically would need help to translate this to working gstreamer pipeline.

ffmpeg -rtsp_transport tcp -i rtsp://ip:port/h264.sdp \
-an -c:v h264 -profile:v baseline -preset ultrafast -tune zerolatency -vf "fps=20" -bsf:v h264_mp4toannexb  -f rtp rtp://localhost:8004 \
-vn -acodec copy -f rtp rtp://localhost:8005

Currenlty I have only this:

gst-launch-1.0 rtspsrc location=rtsp://ip:port/h264.sdp latency=0 protocols=tcp ! rtph264depay ! rtph264pay name=pay0 pt=96 ! udpsink port=8004 host=localhost

but this is not transcoding, and by itself the stream is not playable (strangely it works if I start the ffmpeg example to see video and then run gst but this is beyond the point here), so I would like to have x264 transcoding (re-encoding?) to baseline profile with minimum processing power required.

1 Answer 1


Ok I've found the solution that works, it was here all along but needed some changes (link only since it's link to the stackoverflow)

Your Answer

By clicking “Post Your Answer”, you agree to our terms of service and acknowledge you have read our privacy policy.

Not the answer you're looking for? Browse other questions tagged or ask your own question.