sip is such a terrible service when mixed with NAT
I see it ALL the time… life becomes much simple and easy if you create “bridge’s” or “VPN’s” or Routed networks
As a VoIP telco, I have to agree. Don’t expect it to work. Expect niggles, glitches and strange things. Personally, I am sick to the back teeth of telecoms companies installing systems without having the first clue how the Internet works…
That having been said, there is one thing you can do to make ROS work with SIP better, and that is to disable the SIP ALG which (in my experience) makes things ten times worse. Try:
/ip firewall service-port disable sip
at the command line and see if that improves things.
We place small RouterBoards at our customers’ sites and tunnel VoIP traffic over L2TP to our hosted LNS. Removes NAT from the equation completely and also has the advantage that we then have access in to our customers’ networks so we can reprogram phones and resolve problems remotely.
But then we’re that rare breed of telecoms company - one that knows what it’s doing!
The PBX/server they are registering to may be able to flag registrations as from a NATted source. Asterisk, for example, lets you state “nat=yes” in sip.conf in which case the address within the SIP packets is ignored in favour of the source address of the udp IP packet.
If you cannot mark handsets as being from a NATted source, then I am afraid that it simply will not work for you.
We can’t support all ‘out-of-office’ managers. We want IP-phone usage only.
