HI,all my asterisk11 have a domain name with dynamic ipaddress. i want to use asterisk and linphone make p2p call, i open icesupport in sip.conf and rtp.conf, can call through but no voice.
here https://wiki.asterisk.org/wiki/display/~jcolp/ICE,+STUN,+and+TURN+Support i see ICE support is only used for communication between a remote endpoint and Asterisk. so i close icesupport in sip.conf and rtp.conf, but invite from caller have ice paramter,but asterisk modify invite message, delete ice paramter, then pass to callee, so phenomenon is also can call through but no voice. how to make asterisk not modify sdp of invite, or have other method to use asterisk and linphone make p2p call. thanks!!
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