Hello everybody,
I am having trouble to get a VoiP-application working with our VoiP-server.
The VoiP-application is running on our mobiles and the VoiP-server is behind a Mikrotik router (MikroTik RouterOS 6.38.5).
For some reason the audio is not working. However the trunk-connection between Voip-Server and our VoiP-provider does work perfectly.Even if I do not set any port forwards at all.
Service Ports SIP and H323 have been disabled.
What are the correct port forwards to get the audio-in-two-way working?
best regards,
Ramonn.
What ports did you opened?
There are 2 types of flows:
-
phone signaling. If you use SIP protocol is TCP 5060. This flows allows phones to register on the PBX and get features like the extension number assignment.
-
Voice media streaming. In this case the flow are based on a random UDP port preconfigures on the PBX. As an example on FreePBX /Asterisk the defauly range of ports used (and needs to be allowed) is from UDP 10000 to UDP 20000.
Seems to me you have signaling flows allowed but no media port opened.
Sent from my SM-G920I using Tapatalk
Port forwarding for a voip server only caused issues in the past for me.
Did you specify the dst address in the forwarding rules eg
/ip firewall nat add chain=dstnat dst-address=PUBLIC dst-port=5060 protocol=udp action=dstnat to-address=YOURPBX to-ports=5060
A quick fix was to setup a pptp server on the router and connect the mobile devices to the vpn and the use the private ip of the voip server to register.