Kitz Forum
Broadband Related => Voice over IP (VoIP) => Topic started by: bogof on June 16, 2021, 12:02:37 PM
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Bit of background; had ZEN FTTP installed a couple of months ago (documented here: https://forum.kitz.co.uk/index.php/topic,25446.0.html ). Coming from EE VDSL, which I left in place while making the the FTTP install was sound.
The service is working phenomenally well. I had considered keeping the FTTC as backup, but really it's £25 a month I could do with keeping in my pocket, and the FTTC has so many shared points of failure with FTTP that I'm not sure it is a very good fallback option. So fallback will be 3, 4G based (see here: https://forum.kitz.co.uk/index.php/topic,26059.0.html)
I have a few months left on my 18 month EE contract which I was planning on just letting run; however it turns out they work out their cancellation by taking off any discounts you have, and also any charges they will save to Openreach. This makes the cancellation for 4 remaining months only £20. So I'm porting the number to A&A. This should cause a cease on the EE VDSL service. The savings in line rental will pay for my preloaded 4G SIM card.
Once I have the number on VOIP, I planned to add the VOIP service to mine and my wife's mobile phone. Not sure yet if I'll get an adapter so that I can use the VOIP number on the existing DECT handsets in the house. On the one hand, it's nice to have a phone in the house always wired up, on the other hand, extra cost. I guess I could equally get a hard-wired SIP phone / DECT base station. The Zen contract came with a Fritzbox router too which does support some voice capabilities, and I wonder if perhaps it might even be possible to use that just to interface the old DECT phones to SIP without using it as a router (I use Ubiquiti gear for most of the networking in the house.
Timeline of the port (will update):
14/06/2021 - Went on A&A VOIP page and arranged to have a VOIP account, requesting a port in of my number.
15/06/2021 - Email from a person at A&A confirming the account details for the number being transferred in (I had confirmed during application I thought, perhaps they are just thorough).
16/06/2021 - Email from same person at A&A confirming port in date / time as 22/06/21 10am
22/06/2021 - 10.05am - VOIP incoming calls and voicemail appear to be working fine from my landline number to Android device on office wifi. VDSL ceased service around same time.
I did try using my account details on my mobile phone for SIP setup and they didn't work yet. I don't know if this is because they don't work until the inbound number is attached, or whether there is some other networking issue. I will dig into this a bit when I have a moment, as I would like to ideally know the SIP is up and running before the number ports so we don't have a loss of service on the number. Any thoughts on this?
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Keeping an eye on this. By the way, for testing we used to use a free SIP softphone called X-Lite, now Bria Solo. That was handy for checking account settings and credentials.
Personally I don't like analogue adapters, they seem to be the worst of both analogue or digital worlds for example you may not get CLI properly displayed. At home we use a Gigaset N300A which worked with our existing DECT handsets, and supports analogue line as well as SIP services.
Having said that we now get hardly any landline calls, and make most outbound call via mobile making them effectively free. I generally give out the landline number to people I don't want to have calling the mobile.
Regarding the AA SIP service, when we went SIP there were a couple of points of concern. Firstly they say they don't support NAT, and secondly they specified that the media could be sourced on almost any originating IP address (in contrast to say Gamma where there is a dedicated media address). This latter point might or might be an issue depending on how you manage your firewall, and whether your SIP equipment cooperates in a firewall friendly way.
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Keeping an eye on this. By the way, for testing we used to use a free SIP softphone called X-Lite, now Bria Solo. That was handy for checking account settings and credentials.
Personally I don't like analogue adapters, they seem to be the worst of both analogue or digital worlds for example you may not get CLI properly displayed. At home we use a Gigaset N300A which worked with our existing DECT handsets, and supports analogue line as well as SIP services.
Having said that we now get hardly any landline calls, and make most outbound call via mobile making them effectively free. I generally give out the landline number to people I don't want to have calling the mobile.
Regarding the AA SIP service, when we went SIP there were a couple of points of concern. Firstly they say they don't support NAT, and secondly they specified that the media could be sourced on almost any originating IP address (in contrast to say Gamma where there is a dedicated media address). This latter point might or might be an issue depending on how you manage your firewall, and whether your SIP equipment cooperates in a firewall friendly way.
Thanks for the reply and the pointer to the softphone.
Similarly, we hardly ever get calls on the landline, mostly they are sales. I'm tempted, in a way, to leave the SIP number just going to a voicemail permanently, with the voicemail being emailed to myself and the wife, and then we just call back whoever called. Sometimes the wife has friends who insist on calling landline as they have limited mobile minutes and some free minutes package on the landline, though, so I guess it would be useful to work out how to get it working.
On the NAT point - is that a service provider related issue? I thought SIP and NAT was all about firewall setup at my network, not something related to A&A's service? Quite a lot of internet connections these days are behind the likes of CGNAT so this would seem perhaps to be a pretty big limitation.
I don't understand as yet about media address matters, I will have to do some reading up on this. Thanks for alerting me to the possible issues though.
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Regarding NAT the issue is that a lot of SIP headers contain the original IP address of the device inside your network, particularly the SDP headers that tell the other end where to send their media, that's the "c" header giving IP address and port that your device is listening on. There are some other headers in the SIP itself but these are less often an issue (but we did see one provider looking at the Contact header".
Basic NAT as we all know it will re-write the source IP address and port, leaving any IP addresses mentioned in the payload untouched.
Some services are happy with this, and use their own techniques to work out what's going on. Some require you to enable SIP Inspection or SIP ALG (terminology varies with the equipment). And in some cases these conflict with certain call behaviour.
In terms of A&A they did not say it wouldn't work, but they advise against it.
The issue with media source addresses is that in the worse case you would need to have a pretty huge hole in your firewall to allow all their potential source addresses in. However depending on the behaviour of the service (which they don't specify) and of your kit this may not be necessary. My Gigaset for example does some things that specifically keep firewall ports open just for the conversations in use.
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Interesting, as I had noticed VoIP will not failover on my Gigaset to 4G when my DSL goes down. I assume its probably something to do with this? (as my 4G is double-NAT)
I do have the router with a static IP and set as the DMZ with SIP ALG enabled on the Huawei B535-232 but not sure it actually does anything on a DMZ configuration. Not even sure Voxi issue a public IP.
Ah seems I had the wrong MAC address set for the router, that won't help as it was being issued a random IP from the pool.
[UPDATE]
Can confirm, Voxi is giving out a private IP, apparently Vodafone as a whole have always done this and have no option to get a publc IP. :'(
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SIP may work over CGNAT, depending on the carrier, the equipment and the ITSP. I found it slightly flakey over Smarty, maybe it's the address changes rather than the CGNAT itself. Regarding failover I wonder if your gear is hanging onto the original NAT even though the route is changed. That's what I found mine was doing, because the Gigaset keeps polling this keeps the NAT entry alive. Nowadays my main router just does the routing, NAT isn't done until it reaches the ADSL or the LTE router.
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It shouldn't be, I have Flush all states when a gateway goes down enabled as I also use Do not create rules when gateway is down so I have more control over how failover works, rather than just relying on Gateway Groups.
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Speaking of firewalling I am dealing with an issue that has arisen again with a connection to "Gradwell". Gradwell operate on the principle that your equipment sends its signalling to a defined destination within Gradwell for your outgoing calls. So far so good. However for incoming calls Gradwell sources their call setup messages on a wide range of addresses within a whole page worth of prefixes and lists. So no sort of stateful firewall can cope by default, your firewall needs to be configured to permit unsolicited SIP from any of the addresses on this list ..
https://my.gradwell.com/kA04G0000000DAO?name=what-ip-addresses-may-gradwell-send-voip-traffic-from
To make matters worse the list is subject to change, with no guarantee they will update that web page. So incoming calls are liable to stop working at any time, with some detective work needed to try and figure out what new addresses are being used. This is exactly what was reported by this particular user today. The only other solution would be for the customer to leave their equipment open to unsolicited call attempts from anywhere on the Internet.
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I've not even looked into incoming calls yet, POTS based calls seem to get routed fine, I presume this is only if you want pure IP to IP calls?
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I did try using my account details on my mobile phone for SIP setup and they didn't work yet. I don't know if this is because they don't work until the inbound number is attached, or whether there is some other networking issue. I will dig into this a bit when I have a moment, as I would like to ideally know the SIP is up and running before the number ports so we don't have a loss of service on the number. Any thoughts on this?
When I ported my number in to AA I also tried to set up my phone before the port was complete but it didn't work - was fine once the number had ported though, so it looks like they just don't activate the account until they have the number.
I have been using a Gigaset N300 IP with the service behind NAT just fine, I haven't enabled SIP ALG or anything like that.
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When I ported my number in to AA I also tried to set up my phone before the port was complete but it didn't work - was fine once the number had ported though, so it looks like the just don't activate the account until they have the number.
I have been using a Gigaset N300 IP with the service behind NAT just fine, I haven't enabled SIP ALG or anything like that.
Thanks, that's reassuring. I can see the service appears to have an install date of 22/06/21, I guess it makes sense to only provision the service once the number is transferred in and working.
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Interestingly though the port is scheduled I have had no "sorry you are leaving" etc from EE as a result of the port request. You would think something would come through their various retention systems.
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Well, that was all surprisingly pain free. At 10.05 I attempted to connect my Android mobile phone using the inbuilt SIP client to my SIP account and it connected first time (over my NAT'd office Wifi, running on Ubiquiti Unifi USG3 with more or less default settings). A few minutes later I checked and the VDSL connection is no longer up it seems (TBB ping monitor is now failing). So it appears the cease has rippled through EE's systems.
Nice one!
We will see how well it works on my mobile, and I'll probably look to set up something for the house as I did like us having phones within reach on each floor.
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I've managed to abuse the Zen-provided Fritzbox 7530 into being a client device on my home network, serving as a DECT SIP base station. The 3 cheap old DECT handsets from a Siemens Gigaset POTS base station set have connected up just fine to the Fritzbox. So it seems like I've managed to get a 3 handset DECT SIP setup for nowt! :) :)
Seems to work well. The Fritzbox is behind the router in the house, so it is being NAT'd. Things go a bit wrong if my mobile has the SIP client enabled and connects to the home wifi network - the phones on the Fritzbox stop receiving incoming calls. Not sure if it is possible to get them all able to receive incoming SIP calls if connected to the home network.
Will have to do a bit more reading and playing.
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That's weird, I thought the whole point with SIP is you can effectively login on as many clients as you want and they should all receive calls.
What are you using for SIP on the mobile? Last I tried all the SIP apps were horrible.
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That's weird, I thought the whole point with SIP is you can effectively login on as many clients as you want and they should all receive calls.
What are you using for SIP on the mobile? Last I tried all the SIP apps were horrible.
Yes, but there are problems with several clients behind one NAT gateway, so there are several ways around it it seems. Just working out if I care enough to make it work...
Android has a built in SIP client which you set up in the Calls app. Seems to work well.
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Yes, but there are problems with several clients behind one NAT gateway, so there are several ways around it it seems. Just working out if I care enough to make it work...
Android has a built in SIP client which you set up in the Calls app. Seems to work well.
I did wonder if that was a thing now, not checked in many years.
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As a quick check you could see if your mobile can register without upsetting the home system, using 4G instead of your home wireless. Assuming that works, another thing to try would be to change the source port on your Fritz box to something other that 5060. I am not fully convinced that all SIP services necessarily support multiple concurrent logins,
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I just tested at home and I can register a softphone to my SIP Gate account at the same time as the Gigaset. When I called the number, both rang.
By the way I need to revoke my recommendation for Bria as a softphone. Although in principle it's still free you have to sign up for a trial before you can download it. My new test softphone is "Micro SIP" which is available as a portable application for Windows, you don't even need to install it.
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As a quick check you could see if your mobile can register without upsetting the home system, using 4G instead of your home wireless. Assuming that works, another thing to try would be to change the source port on your Fritz box to something other that 5060. I am not fully convinced that all SIP services necessarily support multiple concurrent logins,
The AAISP control panel shows concurrent logins from my office softphone at another site (3CX) and the Fritzbox at home, and they both ring concurrently to incoming calls, so I think they support multiple logins fine. As far as I can see the Fritzbox SIP client doesn't allow the source port to be changed, and neither does 3CX, though maybe I could do this in the router firewall instead. Not sure.
I'm hesitant to use random SIP clients - they're obviously a vector for expensive mischief, though I've locked down the IP addresses AAISP will allow SIP calls on my account to originate from to my two fixed IP addresses. I'm using 3CX, which was a free download. They are a significantly sized business in the UK with good track record of telephony services.
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My point was that I had two SIP endpoints registered at the same time, through the same Internet connection. So if that doesn't work for you that's probably not because of some inherent limit in SIP, but maybe something to do with your firewall or maybe the SIP service. Does your Fritz box original its SIP from your inside LAN, and then apply NAT, or does it "grab" the outside IP address (so to speak)? It's probably also a scenario where SIP inspection/ALG would be harmful, on hosted systems we've found only one phone can register until the ALG is disabled.
My test was using two devices on the inside LAN both going through the same NAT process to the same service, NAT done by a Zyxel router with ALG disabled.
I agree about random softphones, I mention that Micro SIP only for test purposes, and a portable application that doesn't actually install anything into Windows reduces any risk.
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I also confirm that Sipgate supports more than 1 phone with the same telephone number at the same time. I have been doing this for years in my home. The Sipgate help pages cover the use of different ports for each sip phone ( or ATA box ) to prevent conflicts when used on the same network. It is an advantage to enable port forwarding on the sip ports used, to improve reliability.
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My point was that I had two SIP endpoints registered at the same time, through the same Internet connection.
Had you manually configured them on separate ports?
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No, just left them as their defaults. I can see Gigaset uses 5060 as its source port. MicroSIP uses a random source port by default, but it still works (at the same time as the Gigaset) if I manually configure it to use 5060 as well.
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I also confirm that Sipgate supports more than 1 phone with the same telephone number at the same time. I have been doing this for years in my home. The Sipgate help pages cover the use of different ports for each sip phone ( or ATA box ) to prevent conflicts when used on the same network. It is an advantage to enable port forwarding on the sip ports used, to improve reliability.
With the Gigaset you shouldn't need to enable inbound NAT (port forwarding) assuming a normal stateful firewall. The Gigaset sends out keep-alives sufficiently frequently to keep its particular port open. I imagine any handset intended for this sort of service will do the same. Consider in a hosted environment you may have many phones on the LAN, all on DHCP addresses, keeping inbound mappings configured on the firewall would be impossible.
I took Wireshark traces of various operations with a Gigaset 300 a while ago. I can look them out.
But if you find your device requires a static inbound port mapped, I'd be more interested in what exactly it is doing.
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With my old G/S 486 ATAs, if the router got powered down, the phones would not work properly - voice transmission wise - unless a call had been made prior to an incoming call. Fine, if you knew that the router had lost power at some point, but if you were not around when the break took place, your phone calls would not work properly until the ATAs were restarted again. Port forwarding seems to get over this problem. Maybe my more recent ATAs would not suffer in the same way.
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It sounds like your ATA doesn't refresh it's registration (or not often enough), but your firewall is hanging onto that NAT entry long enough to compensate. After rebooting the router the firewall won't allow SIP in until the ATA re-registers or does something else to trigger the NAT and firewall entry.
In comparison the Gigaset N300 registers with a 600 second expiry, but sends a small UDP packet every 20 seconds which keeps the NAT and firewall inspection alive.
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Thank you for the reply, but I have checked that all my ATA settings are correct as per the Sipgate site. But since carrying out port forwarding on my router, even if I switch everything off together, then switch everything on again, the Sipgate lines work normally again without my doing anything.