I did a code review of all the steps based on your last email and identified that I missed appending the video format for read and write.
ast_channel_set_writeformat(chan, fmt_video); ast_channel_set_readformat(chan, fmt_video); So now I can reflect my audio and video for testing purposes Audio reflector gst-launch-1.0 -v udpsrc port=$1 caps="application/x-rtp, media=audio, encoding-name=PCMU, clock-rate=8000" ! udpsink host=127.0.0.1 port=$2 Video reflector gst-launch-1.0 -v udpsrc port=$1 caps="application/x-rtp, clock-rate=90000, media=video, encoding-name=H264" ! udpsink host=127.0.0.1 port=$2 Thanks for letting me talk this out. Now to work on my crash when I hang up. Michael -- _____________________________________________________________________ -- 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