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.