[asterisk-dev] ast_rtp_engine api

[Michael Blake]

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