[asterisk-biz] asterisk linohone (p2p call between endpoints)

[Mitul Limbani]

Try removing nat=yes
 On Aug 28, 2013 1:19 PM, "王鹏" <voipwangpeng at 126.com> wrote:

>
> 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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