If you are looking for SIP based calling, they SIP and RTSP can be used.
gst-launch-1.0 videotestsrc is-live=true ! x264enc tune=4 key-int-max=60 !
h264parse ! rtph264pay ! udpsink clients=<multicastIp:port>
multicast-iface=<iface>
--
Sent from:
http://gstreamer-devel.966125.n4.nabble.com/_______________________________________________
gstreamer-devel mailing list
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel