It’s made for audio. What exactly are you trying to achieve?
As for mapping… that’s completely in your implementation. You have to allocate a UDP port on your side for the audio, so that’s one way of doing it. Or using the source IP address/port of Asterisk…
okay got it if the rtp server runs on the same machine as the asterisk the port’s will be same
i can manage from there thanks.
so just wanted to send a hang-up signal back if I have to end the call.