Am using rb532 indoor board with v2.9.42 OS working fine except Sip to Sip calls are not working on Linksys PAP2 v1 firmware 3.1.22. When i call PAPB from PAPA, i cant hear the PAPB but PAPB can hear me but faintly.
I’ve enable NAT mapping & NAT support on the Linksys and now i can make SIP 2 SIP calls fine but the sound quality is terrible. I suspect the NAT Firewall on the RouterOS is affecting this. How can i disable the firewall or better still enable this ports on routerOS? Is the beta v3 OS better in handling SIP calls?
SIP signalling ports (UDP) = 5060 - 5061
DNS port (UDP) = 53
TFTP port (UDP) = 69
RTP/RTCP ports (UDP) = 10000 - 30000
Wait a minute … one to one nat? You typically dont even need a SIP helper if you have 1-to-1 nat because all ports are forwarded that way. Really ? So what’s the SIP helper doing then, just changing the private IP for the public one in the udp/5060 payload ? I thought the SIP helper was supposed to enable connection tracking for the RTP/RTSP which runs on other ports as well, similiar to the FTP helper.
I'm using asterisk server (192.168.58.x) behind Mikrotik NAT (192.168.1.1). (asterisk has local voip clients on internal - another ethernet - interface, on 192.168.1.0/24)
udp/5060 and udp/10000-20000 forwarded to asterisk
Before disabling the firewall/services/sip:
Remote clients (coming from internet thru mikrotik) can talk to local clients, asterisk, and call call out, using landline telephony interface.
Voip provider can be used by local phones.
Remote clients cannot use voip provider (no audio -RTP- at all, in any direction!)
After disabling the firewall/services/sip:
Remote clients can do anything as before, plus can use Voip provider!
ps: when using ROS 3.6 or 3.8 (this board originally come with any of these) Asterisk cannot even registery to voip provider (until upgrade to 3.10). (i've not tried to disable SIP service)