Mikrotik RouterOS x86 5.4.

One provider + LAN. i tried to configure srcnat - it's works fine. Turn on uPnP и SIP helper. But my sip softphone doesn't work. But if the port forwarding 5060 and 10000-20000 to my PC - it works. I need to run softphones for each of the 50 PC . How do I get it?

Unfortunately, I have yet to see any kind of router-based SIP helper work 100% correctly. Unless it's actually acting as some kind of Proxy, rather than trying to modify the SIP packets on the fly.

I would actually suggest:
-Disable the SIP protocol in the firewall all together.
-Let the SIP server you're trying to connect to handle the NAT traversal. Most server-side NAT traversal implementations these days do a pretty good job. (If you're connecting to an Asterisk box of some kind-- you should be able to enable NAT support on the SIP peer.)

If you run into issues where it works initially, but stops being able to make/receive calls after awhile, force the registration frequency to something really short like 60 seconds. This creates a little more load on the SIP server, but should keep the NAT connection active.

Is it registration and call control you're having trouble with or media/audio? (or both?)

if you turn on SIP helper and NOT forwards anything, it should work (since 3.10 i think) especially, if the SIP client (softphone or ATA) support SIP Nat. (in most times, the SIP endpoint should support NAT'ed SIP as well)

re-Registering in 60 seconds in NOT a gentle way, SIP ping is used for that. (SIP proxy like OpenSer aka Kamailio support pinging only nat'ed hosts) The main difference is Register is not only generating load on SIP proxy, but in authentication subsystem too.
So, do NOT waste your resources until it's not really must.

Hi,

I am running a wisp down here in South Africa. We had a problem getting our sip phones working on our network and ended up installing an asterisk server. We then had the asterisk server establish the links to the sip provider ie a trunk for each sip connection and the sip phones connect to the asterisk server. The only settings we then added to our Mikrotik Routers was queue prioritization for voip traffic. Works like a charm. Also when our clients phone each other on the network they use their internal number so that decreases the traffic out to the internet. We had trouble with one way audio and sometimes no audio before using asterisk, but now everything works.

Please can you drop me an email sometime, have been dealing with similar issues as yourself. cheers anthony@zeane.com