Gstreamer 1.0 rtspsrc to rtp (audio and video)

Question

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.

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)