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) ??
thanks in advance!
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[Linphone-developers] a linpone-rtp problem.,
Tony<=