Incoming Calls Problem With SNOM D120 VOIP Phone

[#1]

Hi,

I've recently moved to Full Fibre and have signed up with A&A as a VOIP Service Provider. I've purchased a SNOM D120 phone from them and have got outgoing calls working fine.

Still having trouble getting Incoming Calls to work. I'm using the PlusNet-supplied Hub Two router.

After making an Outgoing Call, Incoming Calls work. From my (limited!) understanding of NAT, for a certain period of time after the Outgoing Call was made, the Hub Two router NAT algorithm will think the network traffic for the incoming was related in some way to that Outgoing Call just made, and let it through.

The SNOM D120 phone itself has a SIP Trace - from that I can see that it is sending an SIP "REGISTER" message about every 30 minutes.

I've just tried setting up Port Forwarding of Port 5060 on UDP+TCP protocol for the SNOM D120 IP address (configured the Hub Two router to give this a fixed IP address), but still no luck with incoming calls.

Anybody got any experience of this particular combination of Hub Two Router + SNOM D120 phone + A&A VOIP Service? Or even some general advice of what I might be missing. From my understanding the "every 30 minutes" keep alive REGISTER SIP message from the phone may be too infrequent to persuade the NAT algorithm to let the incoming calls through, but that this "keep alive" mechanism possibly is not required at all if the correct Port Forwarding is set up.

Any help much appreciated! - I am Software Engineer by trade (but not a Network Engineer!) - but this does seem quite fiddly to get to work.

Thanks.

[#2]

@eclipse it does sound like a NAT issue

From my understanding the "every 30 minutes" keep alive REGISTER SIP message from the phone may be too infrequent to persuade the NAT algorithm to let the incoming calls through, but that this "keep alive" mechanism possibly is not required at all if the correct Port Forwarding is set up.

Correct, it is too infrequent, in order to keep the NAT pinhole open , any keepalive typically need to be < 30 Seconds.

Port forwarding would eliminate the need BUT it needs to forward the correct port.

I've just tried setting up Port Forwarding of Port 5060 on UDP+TCP protocol for the SNOM D120 IP address (configured the Hub Two router to give this a fixed IP address), but still no luck with incoming calls.

Forwarding port 5060 isnt going to work, since that's not the port that will be presented on the public interface. Whilst the D120 may be using port 5060 , once its been translated by NAT, it will be a diferent port.

You really need to configure keepalive and it does seem like the D120 has that facility. The manual is not very clear but from what I can see the setting is in the Identity section , https://service.snom.com/display/wiki/keepalive_interval. Set this to 20 and remove any port forwarding.

NB make sure that the SIP ALG is disabled on the Hub 2, it IS by default .

Hope that helps

Let us know how that goes

[#3]

@eclipse have a look at page 69 in the manual, there's a NAT tab in the Identity settings, looks like it should be there.

Note you will need to be in 'admin mode' to change those settings , see page 33

[#4]

@eclipse did you manage to find the keepalive setting ? and did that fix the problem ?

[#5]

I found the a setting "Keepalive Interval" in a section titled "NAT Identity Settings" in the "NAT" tab of "Configuration Identity 1" settings page (all this done through the web interface of the SNOM D120 phone).

The complete list of settings in the NAT Identity Settings:

  STUN Server (IP-Address:Port) ... was already set to stun.a.net.uk

  STUN Interval (s) ... blank

  Keepalive Interval (s) ... blank

  Number of Initial Keep-Alice's on RTP Port ... blank

- The "Keepalive Interval (s)" field was initially blank, I set it to 20, clicked "Apply" 

- Web page displayed an message "Some settings are not yet stored permanently" and clicked the "Save" button that was offered.

The web interface has a "SIP Trace" showing a log of SIP messages.

The SIP log is still showing SIP REGISTER messages being sent out to the VOIP provider about every 30 minutes.

But there have been some successful incoming calls, quite a bit of time after the last 30 minute SIP REGISTER message, which makes me wonder whether the 20 second keep alive is active, but is not showing in the SIP Trace log for some reason?

------------------

For example this morning:

09:45 30 minute SIP REGISTER message

10:05 Incoming call successfully received & started (a real call from my sister in France!)

10:16 30 minute SIP REGISTER message

10:28 above incoming call finished

10:46 30 minute SIP REGISTER message

11:42 Incoming call successfully received (a test call from my mobile)

-----------------------------

(I currently still have the Port Forwarding active as described in my previous post - but plan to remove that to check whether the KeepAlive is sufficient). I think I understand what you were trying to describe about this not working - even though there is a port forwarding, the SIP traffic from A&A on Port 5060 wouldn't know the internal (home side of the NAT) IP address of the SNOM D120 phone - although with NAT algorithms seeming to be mostly heuristically based, if that was the only internal device for which Port 5060 had Port Forwarding set up for, I guess a NAT algorithm could decide that the SNOM D120 phone is very likely to be the intended recipient of the SIP message.

I also did notice that the SIP REGISTER messages sent from the SNOM D120 phone to the A&A VOIP provider do contain the internal IP address (house/internal side of the NAT) of the phone in this record: "X-Real-IP: 192.168.1.98"

[#6]

@eclipse 

which makes me wonder whether the 20 second keep alive is active, but is not showing in the SIP Trace log for some reason?

The keepalive packets wont show in the SIP trace, they are not SIP messages as such, they are simply a UDP packet using the same address/port as the SIP server registration and thus will keep the NAT pinhole open so that when a SIP invite ( incoming call ) arrives it gets routed correctly to the SNOM.

I think I understand what you were trying to describe about this not working - even though there is a port forwarding, the SIP traffic from A&A on Port 5060 wouldn't know the internal (home side of the NAT) IP address of the SNOM D120 phone - although with NAT algorithms seeming to be mostly heuristically based, if that was the only internal device for which Port 5060 had Port Forwarding set up for, I guess a NAT algorithm could decide that the SNOM D120 phone is very likely to be the intended recipient of the SIP message.

The problem is that whilst your SNOM may use port 5060 as the source port, when the REGISTER goes out via the router NAT the source port will be changed , so the A & A servers will see a registration from your public IP but with a source port of for example 20000. So when it send the incoming call INVITE it will send it to your public IP but to port 20000. So a port forward of destination port 5060 will not allow that packet.

I also did notice that the SIP REGISTER messages sent from the SNOM D120 phone to the A&A VOIP provider do contain the internal IP address (house/internal side of the NAT) of the phone in this record: "X-Real-IP: 192.168.1.98"

That's to be expected and is the basic problem with SIP behind NAT!. In theory the address in the register packet is the address to which INVITEs should be sent BUT of course when behind NAT its a private address and so is non-reachable externally. Fortunately most SIP servers are NAT aware and ignore the address in the REGISTER and actually use the address from which the REGISTER came i.e your public IP!.

From your examples, it would seem that the keepalive is working. You've received calls 15 min after the previous register, so the NAT pinhole must still be open and whilst I'm not certain what the NAT timeout is on the Hub 2 , I would guess its around 30 seconds