[SR-Users] kamailio + asterisk + webrtc

[Kenan Kocaerkek]

Hi Arish,

Can you provide asterisk's sip.conf and example SDP messages related to
your calls initiated from outside.

You must use advertised address in kamailio config as below.
listen=udp:192.168.1.x:5060 advertise 182.70.xx.yy:5060


arish haque <arish.haq at gmail.com>, 22 Eki 2018 Pzt, 12:40 tarihinde şunu
yazdı:

> Hi all,
> I am using kamailio as sip server and asterisk as media server behind
> pf-sense
> firewall (NAT) with a public ip.
> Sip signalling as well as rtp packets are flowing correctly when the
> endpoints are within
> a network. But when trying from outside network sip signalling is working
> perfectly
> but there is no RTP.
>
> Public ip behind pfsense -> 182.70.xx.yy
> kamailio+rtpengine and asterisk servers are on private ips --> 192.168.1.x
> kamailio ip - 192.168.1.x
> asterisk ip - 192.168.1.y
>
> Please, feel free to ask for more information.
>
> Thanks & Regards,
> Arish Haque
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