[SR-Users] kamailio + asterisk + webrtc

[Fred Posner]

PFsense has no inherent problems for VoIP. Make sure you’re using an advertised public address for SIP and RTP and also make sure you’re forwarding RTP port range from outside to RTP engine. 

Fred Posner
direct: +1 (224) 334-FRED (3733)

> On Oct 22, 2018, at 5:40 AM, arish haque <arish.haq at gmail.com> wrote:
> 
> 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