Problem with a VoIP phone behind NAT - disabling FortiGate

Aug 21, 2016 Using SIP Devices behind NAT – The Smartvox Knowledgebase The same applies to SIP servers behind NAT – e.g. Asterisk – where you can specify the range of port numbers to be used for media sessions. Once you have defined that range of port numbers, you simply have to set the firewall/NAT device to forward that range of ports to the IP phone or server. This should work provided the external IP Asterisk Forums • View topic - Asterisk 10 and nat=yes Mar 31, 2013 RTP / NAT Question ( Pjsip ) - Asterisk FAQs Mar 02, 2016

OpenSIPS - Configuration and Integration with Asterisk

Solved: Disable NAT on SIP payload - breaks ICE - Check

Securing Your Asterisk VoIP Server with IPTables

Sep 05, 2017 Troubleshoot and solve one-way audio in VoIP by - Asterisk This should be considered good basic LAN network design anyway and multiple instances of NAT will cause problems both in VoIP and elsewhere. One-way audio can occur in either direction, however in-bound audio failure (lack of audio from the outside caller reaching the inside network (LAN) phone) is probably the most common. NAT and Firewall Traversal Recommendation – OnSIP Support A NAT router with a built-in SIP ALG can re-write information within the SIP messages (SIP headers and SDP body) making signaling and audio traffic between the client behind NAT … Asterisk trying to send RTP packets to the internal Jan 06, 2013