I've just had a test on linphone app; and came across such a rtp session puzzle! when my call-invite set up, and the rtp session was going on smoothly,I catch the rtp packets through wireshark ,all the rtp packets are from 192.168.1.88 to 91.121.196.132;
I think the 91.121.196.132 is the sip server , I want to now that if rtp packets need to pass by sip server???
why it isn't the two terminal directly go on with rtp communication??? just directly from 192.168.1.88(61.247.89.78:8870) to 192.168.1.40( 61.247.89.78::7887) ??