This can be done with same example https://github.com/centricular/gstwebrtc-demos which is used a reference for this demo. Thanks to the Centricular Team for this support
Please run the webapp and note down the peer id generated, say example, 1232
cd */gst-webrtc-example mkdir cmake-build-debug cd cmake-build-debug cmake .. make
rtsp2webrtc_1_n
Create folder 'mkdir /mnt/av/ ' with write permissions
Or update const variable 'BASE_RECORDING_PATH' in file rtsp_webrtc_1_n.cpp accordingly
./rtsp2webrtc_1_n |SIGNALLING SERVER URL| |PEER ID NOTED FROM BROWSER| |PCAP FILE PATH| |RTP SOURCE IP| |RTP SOURCE PORT|
Example: ./rtsp2webrtc_1_n wss://127.0.0.1:8443 1232 /home/user/pcap_recording_webrtc_stuttering_10_56598.pcap 192.168.0.10 56598
And check logs, video for further analysis