One way RTP traffic
Hoping someone here might have some ideas around this one.
I have set up an Asterisk PBX behind a Pfsense firewall. My internet IP is allocated to the WAN interface of the firewall with the Asterisk box being allocated a local network address.
Soft phones within the local network or connecting to the network via OpenVPN are able to call each other without issues.
The issue is that when a call is made to or received from remote soft phone that has registered with the PBX over the internet (no VPN) both the A and B parties hear dead air.
Looking at a packet capture from the WAN interface of the Firewall (see attached) I can see the RTP stream from the internal softphone leaving the Firewall but no RTP packets are received from the remote party.
Please note that the issue only happens with RTP packets, SIP packets (Register, Ack etc.) are sent and received without any issues.
Thanks in advance
[One way RTP.pcap](/public/imported_attachments/1/One way RTP.pcap)
Rtp needs open ports for the return traffic. Rtp.conf on the asterisk server is where you can define a range of ports to use such as UDP 10000-20000 . Then in pfsense create a WAN rule allowing the range in to your asterisk server. This is in addition to 5060 for the SIP session and the NAT rule. If both sides are behind NAT you will need a STUN server to assist with the connection I believe.