VoIP Network FAQs
Question 1) How do I disable SIP ALG in my Technicolor TG588v V2/TG589Vac?
Answer) If you have a older version of a Technicolor (TG582n or TG589vn v3) or a Thomson Gateway router then please click here
To turn off SIP ALG in the Technicolor TG588v V2, follow these simple steps:
You’ll need to enter a special address in order to access the SIP ALG options on your Technicolor TG588v V2, You enter the router’s IP address (default is usually 192.168.1.1) and then at the end of the address enter /?debug=1 Make sure to log in as Administrator, with the default credentials being:
Username: admin
Password: admin
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SIP ALG has now been successfully disabled on your router!
If you need any further help today, please don't hesitate to contact our friendly support team on +443300437543 or by email!
Question 2) How do I disable SIP ALG in my Technicolor TG582n/TG589/TG585?
Answer) The following router models all feature "SIP ALG", and you will need to disable this for a sound and robust VoIP service from any VoIP provider: Technicolor TG582n PRO, Technicolor TG582n SB, Technicolor TG589, Thomson TG585 v7 and TG585 v8.
If you have a newer version of these routers, such as a TTG588v V2 or TG589Vac, you can view our FAQ on those routers here:
How do I disable SIP ALG in my Technicolor TG588v V2/TG589Vac?
To disable SIP ALG on these routers you will need to do the following on a Windows computer:
1) Press the Windows Start button and in the search box type in cmd and hit Enter
2) Type in telnet 192.168.1.254 and hit Enter. If this doesn't work, then you don't have Telnet installed and will need to go to Start → Control Panel → Programs → Programs and Features → Turn Windows Features on or off → Ensure TELNET CLIENT is checked and click OK.
For a Mac, you won't need to turn on any features like in Windows, simply open the Terminal Application and type telnet 192.168.1.254 and hitting enter:
3) When prompted, the default username is Administrator (case sensitive) and the default password is either blank or the serial number of the router (case sensitive):
Note: When you enter the information, no confirmation message will appear to let you know that SIP ALG has been turned off. Instead, if the command is typed correctly, a new line will appear beneath the command as seen below:
4) Type in "connection unbind application=SIP(TCP) port=5060" (without the quotation marks) and press Enter. If you get a 'unknown command' message returned once you press enter, then type in: "connection unbind application=SIP port=5060" (without the quotation marks)
5) Type in "saveall" and press Enter to apply and make this setting permanent.
6) Type in "exit" and press Enter to exit.
If you need any further help today, please don't hesitate to contact our friendly support team on +443300437543 or by email!
Question 3) How do I set up a Draytek with a Virgin Media Superhub?
Answer) To set up a Draytek with a Virgin Superhub, you'll first need to put the Virgin Superhub into modem mode. To do so, follow this link: How do I put my Virgin Media Hub into modem mode?
Next, head into the web configuration for the Draytek, which can be accessed via the web address 192.168.1.1, unless specified differently by your home or business network, using the default username and password admin to log in.
Navigate to WAN > General down the left-hand side, and check that the physical mode/type is set to Ethernet/Auto-negotiation for the WAN port you used - in the screenshot below, we have used WAN 2:
Next, go to Wan > Internet Access down the left-hand side, and set the Access Mode for the WAN port you used to Static or Dynamic IP. In the screenshot below we used WAN 2 again:
If the above is all correct and saved, reboot your Super Hub; it'll take around a minute for your Draytek to grab an IP address from your Virgin SuperHub once you've rebooted it, however after a short delay you should have internet.
Once connected, we recommend setting up Quality of Service on your Draytek in order to manage your bandwidth efficiently. Our FAQ can guide you through this process.
If you need any further help today, please don't hesitate to contact our friendly support team on +443300437543 or by email!
Question 4) How do I set up a Draytek with an ADSL connection on Sky/TalkTalk/BT/Plusnet?
Answer) ADSL refers to any router that is connected via the phone line. In most cases, this is a standard BT, Plusnet, TalkTalk or Sky router.
To set up a Draytek with ADSL, you'll first need to get your ADSL credentials from your current provider. This will be in the form of an e-mail address or password and will be needed in order to power your Draytek with the internet.
Most providers will give these to you when asked but Sky will take some persuading because of their terms and conditions advising against using a 3rd party router.
You'll also need to make sure that the ethernet cable is plugged into the correct port (WAN2) as seen below, and your DSL (phone line) cable is plugged into the DSL port on your Draytek:
You would then need to head into the web configuration for the Draytek, which can be accessed via the web address of 192.168.1.1, unless specified differently by your home or business network, using the default username and password of admin to log in.
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After configuring WAN1, select the Quick Start Wizard once more, and then select WAN2 as seen below and follow the steps!
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Once you have clicked OK, reboot your Draytek and it'll take around 5-10 minutes for your Draytek to grab an IP address and then you'll have internet!
Once connected, we recommend setting up Quality of Service on your Draytek in order to manage your bandwidth efficiently. Our FAQ can guide you through this process.
If you need any further help today, please don't hesitate to contact our friendly support team on +443300437543 or by email!
Question 5) How much bandwidth do I need to make a VoIP call work?
Answer) VoIP uses your Internet data connection so it is important you have adequate bandwidth. You can save bandwidth by using more efficient codecs (Audio Compressors) in the settings of your hardware of softphone, but the most common default standard is a codec called g.711. This gives 64 Kbps sound quality - the exceptional quality we think you'll want! Alternative codecs like g.722 can reduce bandwidth requirements over the industry standard g.711 by as much as 20%, but reduce the call quality to 48 Kbps. Nonetheless, still very adequate for most though.
So let’s base our maths on the more normal g.711 codec for now. With the overheads that VoIP needs to control a call, you'll need just short of 80 Kbps for each part of the call, the outgoing and incoming; the two parts of a conversation. So in bandwidth terms, that means you will need to have at least 80 Kbps of upload speed and at least 80 Kbps of download speed per concurrent phone call. 8% of each leg of a pure uncontended 1Mbps upload/download broadband speed if you like.
In reality, we regularly see VoIP bandwidth usage at substantially less than 80Kbps on each leg of the download/upload as most phone conversations don't have two people talking constantly at each other - they are a conversation and not all the bandwidth is consumed at once. However, this is a max figure that you should budget your calculations on.
Bits and Bytes and in turn Megabits and Megabytes are different, so be careful not to get them confused when seeing what bandwidth you have available. A byte/Megabyte is broadly 8 x bigger than a Bit or Megabit. Broadband/Bandwidth is generally marketed and measured in Megabits so a 1MB file on a pure uncontended 1MB per second download speed will actually take 8 seconds to download in a perfect world, not the 1 second you might first think. For ease with our maths here, a Kbps is a unit of a Megabit, so you don't need to do any conversions, but we wanted to explain our Bits and Bytes well!
Concurrency is key in your calculation. You may have say 30 users, but it's highly unlikely you will see that many on a call at once. In fact, from our stats, we see an average of 1 concurrent call for every 3 users. So for 30 users, we would suggest your bandwidth requirement would be 10 x 80 Kbps (800 Kbps) upload and 10 x 80 Kbps (800 Kbps) download for the large majority of your run rate business.
Finally, we would encourage you to test your actual bandwidth using tools like Speedtest.net. Your actual bandwidth may vary greatly from what you think you buy from your provider at different times of the day and depending on the priority they are giving to your traffic.
Your download speed will likely be substantially more than your upload speed, so upload is the key constraining measurement to take a good look at. I sit right now writing this FAQ from a regular domestic BT fibre connection in a city and am achieving 15 Megabits download and 11 Megabits upload; that’s enough for 137 concurrent calls based on my upload (80 x 137 = 10.96 Megabits) and no other usage for my connection!
If you need any further help today, please don't hesitate to contact our friendly support team on +443300437543 or by email!
Question 6) What can I do if I have a BT Business Hub Version 5 and am having issues making and receiving calls?
Answer) This router has a feature called "SIP ALG", and you will need to disable this for a sound and robust VoIP service from any VoIP provider. Just following the following steps.
1) Make sure you're using a computer that's connected to your network
2) Open a browser (Internet Explorer, Firefox etc) and type 192.168.1.254 or businesshub.home in the address bar. You'll see the BT Business Hub home page.
3) Click on ‘Advanced Settings’
4) Enter your admin password (unless this has been changed, the default admin password can be found on the bottom of the Business Hub 5 or on the plastic pull out on its back).
5) Click ‘Continue to Advanced Settings’
6) Click ‘Firewall’ then ‘Configuration’ and scroll down to ‘Application Layer Gateway’
7) Click Yes next to ‘Disable SIP ALG’ and click ‘Apply’
SIP ALG will now be deactivated on your Business Hub 5.
If you need any further help today, please don't hesitate to contact our friendly support team on +443300437543 or by email!
Question 7) How do I setup LanScan IP scanner for Mac OS?
Answer) LanScan for Mac is a free IP scanner for Mac OS that allows you to scan your local network to see everything operating on it. We like scanners like these, because they help you to easily find and identify the current IP of your hardware phone. Once you know your IP, you can put it into your web browser and login to the phone to configure; much easier than using the tiny menu and keys on the phone.
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If you need any further help today, please don't hesitate to contact our friendly support team on +443300437543 or by email!
Question 8) We are located in a Regus serviced office and I can't get my service to work?
Answer) Regus serviced offices block VoIP traffic by default which means your hardware or software phone can't access any VoIP providers network.
They permit the traffic upon request and will direct you to an online form on their website where you enter the MAC (Media Access Control) address's for each of your devices. Once you input these everything works perfectly!
MAC Address's are the physical address your device has on a network. It's easily identified on most phones as its printed on a sticker you can find somewhere on the handset. For softphones and/or smartphones, you'll need to get the MAC address of your device from the devices user interface.
To find a Windows computer MAC address:
1) Navigate to the "Command Prompt". This can be found in your main menu or by selecting "Run" from the main menu and typing "Cmd".
2) Once at the prompt, type "ipconfig /all" and press enter
3) This will show you all your active network interfaces. Find the correct interface for your connection to the Internet (eg: WiFI Adaptor or Ethernet network card)
4) The MAC address will be shown below the relevant connection following the words "Physical Address"
To find an Apple Mac computer MAC address:
1) Click on the Apple icon in the top left corner and select System Preferences
2) Click on Network
3) Select the interface you are using to connect to the Internet from the left. These are normally shown in green as the active connections.
4) Click on Advanced in the bottom right and then select the Hardware tab.
If you need any further help today, please don't hesitate to contact our friendly support team on +443300437543 or by email!
Question 9) What ports on my firewall and/or network do VoIP calls use?
Answer) VoIP calls consist of three major steps:
1) A SIP registration to the VivaTel Platform so we know you are connected to us and ready to make and receive calls
2) A SIP Invite Conversation when you actually make and or receive a call
3) The actual audio of the call itself which takes place within the “SIP Invite" conversation but using RTP to communicate the audio to and from you and VivaTel.
Steps 1 & 2 take place over UDP or TCP Port 5060 if un-encrypted or 5061 over TCP, if encrypted. Encryption is also known as TLS and you need to activate encryption on both your phone and in the VivaTel Dashboard at Dashboard > My VoIP > SIP Users if its being used.
Step 3 can pretty much use any port you wish. The VivaTel platform will always use a port between 10,000 and 40,000 and your softphone or hardware phone will define which port it wishes to use.
In all cases, the above is detail you shouldn’t really need to know and that’s because each part of VoIP call is initiated by each side. So nothing special needs to be done to your firewall to make a call happen.
If you need any further help today, please don't hesitate to contact our friendly support team on +443300437543 or by email!
Question 10) How do I put my Virgin Media Hitron CGNv4 router into modem mode?
Answer) To put a Virgin Media Business router (hereafter referred to as Hitron) into Modem mode, you can follow the link from Virgin Media's website below:
How do I put my Virgin Media Hitron router into modem mode
The alternative way if you are unsure is to contact Virgin themselves, in which their staff will be able to do it for you!
If you need any further help today, please don't hesitate to contact our friendly support team on +443300437543 or by email!
Question 11) How do I turn off SIP ALG on a Draytek router?
Answer) SIP ALG should be turned off in a Draytek by default. If not, then you can follow the steps below to turn it off:
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If you need any further help today, please don't hesitate to contact our friendly support team on +443300437543 or by email!
Question 12) What is the maximum MTU size I can transmit and receive to SIP traffic if I’m using UDP?
Answer) There are two transport protocols that we use to carry VoIP traffic between your network and ours. UDP and TCP/IP. UDP is the most common standard for VoIP but is limited in capability.
The maximum MTU (Maximum Transmission Unit) size of UDP we can receive un-fragmented is 1460 bytes. Most broadband routers are set to an MTU default of 1454 bytes, so its uncommon you can transmit more than our limits. You can send us larger sizes, but that would mean our receipt is fragmented and we won't receive all the data we may need to make your call work.
The data you send us with each call contains information about how our platform can communicate with you, who you are calling, how you are presenting your call, your settings such as the codecs you support and a heap of other information we need to make a call function.
If your MTU size is larger than 1460 bytes, it can be reduced by removing some of the optional needless information you may transmit to us. We commonly see users limiting the number of Codecs in their device's settings to the four main codecs of OPUS, G.722, G.711A (Also referred to as PCMA) and G.711U(Also referred to as PCMU) in that order.
Whilst we do fully appreciate network engineers will often prefer to use UDP for VoIP, MTU size is a limited and repetitive issue we see causing communication problems. TCP/IP has no material limit of concern, so its worth considering switching your devices from UDP to TCP/IP as their transport.
If you need any further help today, please don't hesitate to contact our friendly support team on +443300437543 or by email!
Question 13) What is the ‘Emergency Services Line Identifier’ and what does it do?
Answer) VoIP is very different to an Analogue phone line as calls can be made to and from the same number from anywhere in the world. This is a fantastic feature, however, can be a little problematic in times of emergency:
With a traditional phone line, you would have one address using that number, which is also connected to the local exchange. This means that, if someone were to call the emergency services, then there is a unique address that can be obtained from that number. With a VoIP line, if the emergency services are called it is harder to ascertain the location of the call as it is made over the internet and can be done from anywhere.
As a result, with every VoIP number, we ask that an Emergency Address be added and assigned to each number, which then registers the address to be used by emergency services in the event of an emergency call.
This is all very helpful, but can still cause problems in situations where multiple users are using the same Caller ID, but from different locations - which user resides at the emergency address given?
Unfortunately, it isn’t as easy as assigning an address to the user itself, as we only ever send out the information about the specific call, and all the ‘User bits’ are dealt with within our platform.
In response, we created the Emergency Services Line Identifier, which allows Users to assign a second, ‘hidden’ Caller ID which changes the P-Asserted-Identity Header for your calls. In English, this means that we will send an underlying number which will be used by the Emergency Services to determine the address in the event of an Emergency Call.
If you would like to use the ‘Emergency Services Line Identifier’ you will need to add an Emergency Address in your Dashboard, under VoIP > Config > Addresses, and then assign it to a Number (which would serve as the Number permanently based at that specific address) at VoIP > Numbers. Once that number has an Emergency Address Registered you can then choose the number as your User’s Emergency Services Line Identifier, which will always send out this underlying number, which associates your user with that number address. You’ll then be free to choose whatever Number to Present (Caller ID) that you like, as it will only have an effect on the number your callee sees!
If you need any further help today, please don't hesitate to contact our friendly support team on +443300437543 or by email!