I want to find what is the gstreamer command line options that will generate a rtp stream that can be served to asterisk
Taking note that asterisk supports h263/h264, This is the command I use for generating and serving the rtp in h263. gst-launch-1.0 -v rtpbin name=rtpbin videotestsrc ! video/x-raw,width=352,height=288 ! avenc_h263 ! rtph263pay pt=34 ! rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! udpsink host=10.1.1.1 port=5002 audiotestsrc samplesperbuffer=1000 ! mulawenc ! rtppcmupay pt=0 max-ptime=20000000 timestamp-offset=0 ! rtpbin.send_rtp_sink_1 rtpbin.send_rtp_src_1 ! udpsink host=10.1.1.1 port=5000 I confirm that this is working. I use this command to receive the rtp as a client. This is working fine and received and displayed as expected. gst-launch -v udpsrc uri=udp://10.1.1.1:5002 ! application/x-rtp,clock-rate=90000,payload=34 ! rtph263depay ! ffdec_h263 ! autovideosink Concluding that the above server command is valid by the fact that it is sent and received successfully in the tests above why then does rtp sent in this format using the same comand options not work with asterisk when. Can any body advise the required spec for asterisk to handle video rtp stream, what am I missing. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
