AVM Fritz!Box used as ATA behind a Mikrotik NAT Router?

RouterOS General
1

Hi! I searched around, but so far I couldn’t find a real solution for my problem. I use a AVM Fritz!Box 7050 as some ATA behind a 532A Mikrotik Router which does the PPPoE and NAT. So far I succeeded that I can make outgoing calls, but incoming works sporadically only and mostly for a limited time - typically 30 seconds.

I don’t really like port forwarding as maybe there will be more than one ATA behind the NAT gateway sooner or later. Is STUN a possible solution? I have the feeling that is works with “Sipgate.de” but not with “1&1”. Any experiences?

Regards from Germany,
Purrfect

2

SIP and NAT go together like cats and water. It just hates it.

STUN would be a good solution, depending on your service.

I know that the SPA2002/PAP2 work very well on our VoIP solution (this is one small occurrence of SIP and NAT cooperating =). We have dozens of them behind a single public IP and many in residential deployment.

I’m told that Vonage and Packet8 work behind NAT, but I do not have any experience. I know that Vonage uses the SPA2002/PAP2 devices but I’ve no idea where Packet8 comes into play technology wise.

3

Thanks! I noticed SIP and NAT don’t like eachother. Hmm… I have been experimenting for quite a while and found strange effects. Even though I’m using STUN sometimes audio goes in one direction only… I also tried port forwarding and it works with mobile phones but not with landlines… sigh I’m running out of ideas.

I read that the new Mikrotik OS 3 will have a better support of SIP but I cannot update my router as its mounted on a roof of a company where I can only access in emergency… I don’t dare to update it remotely as maybe it might lose its config thus locking me out of any further access as it’s a wireless connection only…

Is there a cookbook how to make a Mikrotik 2.9.46 aware of SIP traffic and forward it correctly? I really need this ATA to work soon. It’s ok if I need to have ports opened as long as it works… :frowning:

Thanks in advance,
Purrfect

4

I am by no means an expert, but I don’t believe the router you’re using is relevant to your problem. It is your VoIP equipment.

You’ll also want to note that there is not just one version of SIP. There are many variations of it, even though they all carry the name “SIP”.

You might want to look at getting a block of public IPs from you ISP and just stick them on your SIP devices.

5

About SIP and NAT relations you can read here,
http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions

3.0 version contains SIP helper for SIP traffic, that is being natted.
‘ip firewall service-port’ contains NAT helpers.

6

Thanks. I have been experimenting with the sip helper (v3.0rc13) but I don’t know what it does exactly? Do I need to enter the SIP ports that are needed for my ATA device or does it find out automatically? Even though I tried I couldn’t get a reliable connection to my telephony provider yet, it’s very frustrating. Is there a way to monitor the sip helper perhaps?

I also would like to try forwarding specific ports to my ATA device manually too, but how can I achieve this as my ISP IP address is dynamic only. Thus I cannot simply enter a dstnat rule as it demands a static address (correct?) or is there another way to achieve this?

Regards,
Purrfect