Gstreamer 1.0 rtspsrc to rtp (audio and video) 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.
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)