Question to RTP-Setting

When i make phonecalls and examine the traffic, i can see that the voicepackages are sent to the asterisk-server and the server forwards the packages to the other client.

is there a setting where i can change it? i want the clients communicate to each other directy over rtp. the asterisk-server should only do the sip-part.

i need this for my study. would be nice if anybody could help me with that problem.

directmedia=yes for both parties, combined with a lack of any contraindications for direct media, such as incompatible codecs, enabled features codes, call recording, etc. Either party may reject direct media.

Please note this forum is for discussions, not for questions.

OK Sorry
but thank you for your fast answer :smiley: