This application makes a webrtc video call with jitsi meet signaling. Other end can be any jitsi meet app or web app. It doesn't support a room with more than 2 people.

• Apply patch in gst-patch to respective gstreamer repo and rebuild, reinstall changed .so file
• Clone gst-examples repo from here https://gitlab.freedesktop.org/gstreamer/gst-examples, replace webrtc-sendrecv.c with file from gst-examples folder and rebuild

• Check result of this command first: media-ctl -d /dev/media1 -p If it doesn't output the camera dev topology, then you need to find the correct media dev, and replace it in webrtc-sendrecv.sh
• Run this command: webrtc-sendrecv.sh nick_name room_name This will create a new room or join a room.
• Sometimes it doesn't work, maybe the signaling failed. If you don't see any change after 10 secs then you need to kill it and try again.

• jitsi_websocket.py will connect to jitsi meet server through websocket and exchange xmpp, convert xmpp to sdp, and send/receive sdp to/from webrtc-sendrecv program.
• webrtc-sendrecv use gstreamer webrtcbin. You can modify the pipeline string in code to stream audio/video.

https://www.youtube.com/watch?v=AlmOvYJzIoI

• I changed my focus to other project and not working on this anymore, maybe I will comeback to it someday though.

• This one look well organized: https://github.com/avstack/gst-meet