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We have created a comprehensive page dedicated to Internet telephony related acronyms, terms, and definitions at https://www.kiwivoip.co.nz/acronyms
After purchasing some hardware, if you want to use it for SIP VoIP applications (ie., dialing other VoIP users, or dialing out to the public telephone network), you will need to sign up with a service provider. There are many providers out there offering an array of features – depending on whether you plan on making a lot of PC-Phone calls or just PC-PC calls. Here is a complete comparison of many of the providers.
In order to use voice over IP you require a VoIP client – this can be either software or hardware. Stand-alone IP phones or adapters such as the Snom 300, or Grandstream Handytone 286 do not require any software on your computer in order to operate (in fact, they don’t even require you to have a computer!) They plug into a spare Ethernet port on your router/switch, and operate totally independently from your computer.
Software for use with voice over IP is known as a soft-phone. Instead of using a stand-alone device for VoIP, you can use an application on your computer. You then require a speaker and microphone (or a handset or headset) to connect to your computer. To connect to SIP based VoIP providers, such as the ones you can connect to with the stand-alone hardware, you need to use SIP compatible hardware. Three popular SIP compatible soft-phones are:
Some soft-phones are not SIP compatible, but still fall into the category of a voice over IP application. These also allow voice conversations, but are limited to the proprietary environment they operate in. Examples include the ‘talk’ feature on MSN Messenger, ‘talking’ on AOL Instant Messenger and Skype.
Although it recommended that you have a broadband connection in order to use VoIP applications, VoIP can work just fine over a dialup connection. Remember, that it requires two people to use VoIP, you and the person you’re calling – the quality of the audio in the conversation will be determined by the person with the lower quality connection.
Skype, for example, appears to have good quality over a dialup modem. Whilst SIP based VoIP is affected more by the speed of the connection. The quality of the audio is determined by the codec used (CODer, EnCoder). The codec used for a given conversation is determined by the two parties in the conversation upon initialisation. The ‘best’ codec that both parties can use, will be used. If one party is on a dialup connection, their VoIP application will only ‘offer’ a codec which can manage high compression – which implies, lower quality.
So in short – if your dialup connection is 56K, and you are happy to not use other Internet applications while on the Internet phone, then it is worth giving it a try, even if only just to test it out. In the long term though, you really will need broadband in order to experience good quality calls without any jitters, jumps and so forth. If you are thinking of broadband, contact us for an attractive DSL package to suit your requirements. Whether its for home or business we can sign you up with a package to suit.
Like most Internet applications, VoIP was not designed specifically to work behind network address translators. The reason for this is that VoIP relies heavily on UDP traffic – both incoming and outgoing. That being said, given that most home networks are behind NATs SIP has had features added to it to make it work in these situations. In most cases there is a way to get around your router’s firewall and NAT. Mostly this is accomplished by the end device (your SIP hardware, of soft-phone) and the server it connects to, but sometimes you need to make changes to your router.
The use of STUN and outbound proxies are two common methods of getting around NATs. Using an outbound proxy involves relaying all traffic via a third party. STUN on the other hand is used by the SIP device to discover what sort of router you have and what the public IP address of your router is. Both of these methods are transparent to the user and should not be of much concern.
Some routers that offer stateful packet inspection (SPI) are smart enough to know when you have a SIP device on your local network, and will be able to deal with the traffic quite well, without requiring any special settings or configurations. If you have trouble using VoIP on your home network, it is generally advised to make sure you have the latest firmware running on your router (please consult your router’s manual to find out how to update the firmware). As VoIP gains in popularity, many router manufacturers are making sure their routers are compatible, and often this means releasing firmware updates.
If you are faced with a router which appears to give you endless trouble when trying to use VoIP, a last resort is to set up port forwarding. The first effect of this is ensuring that the firewall does not block traffic on certain ports from coming in to your network. The second effect is to ensure that all traffic coming in on specific ports is forwarded to the IP address of your SIP device. The standard ports used by SIP are 5004 and 5060 for UDP traffic (a few providers use non-standard ports). Please consult your router’s user manual if you are unsure how to set up port forwarding.
For more information on using voice over IP applications behind network address translators, see the VoIP-info Wiki.
VoIP is a new technology, and no-one expects you to be able to set it up all by yourself. Any product sold through kiwivoip.com comes with the assurance that we will help you get it up and running. If that means helping you over the phone, or simply answering some questions over email, we will help you.
In some circumstances, the products will only need to be plugged in, and they will work. But in some cases, extra configuration may be required (for example, if you are using a SIP-unfriendly router). Either way, we will not rest until you make some free phone calls!
Sure! If you want to purchase one of the self contained ATAs (analogue telephone adapters), you just need to make sure your wireless router has Ethernet ports on it – as the ATA will plug into one of those. It does not matter that the rest of your network is wireless. Another option is to simply use a Wifi IP phone that talks directly to your 802.11b/g wireless router/access point.
When you use VoIP, one of a number of encoding algorithms is used – depending on the bandwidth available at each end. The ‘benchmark’ algorithm is the uncompressed one – how standard phone calls are sent along the phone lines. This takes up 64kbps in each direction. Adding overheads – it is approximately 80kbps in each direction. Other algorithms (which are more commonly used with VoIP) are indistinguishable in quality, but add compression (similar to how MP3s manage to compress the music, without affecting the quality much). These encoding algorithms generally reduce the bandwidth (including overheads) to between 12 and 36kbps. In addition to the encoding algorithm, VoIP does ‘silence suppression’. Since for much of the phone call, one (or both) parties are not speaking, it is pointless to take up all that bandwidth transmitting silence – so this part is not transmitted. As a general rule, you might say that 100% of the time, one party isn’t talking (in other words, each person doesn’t talk for 50% of the time). This approximately halves the bandwidth usage.
See The VoIP Calculator for more details.
It does not matter who your ISP is when you use VoIP. As a general rules, VoIP providers do not offer ISP service (and vice versa). It it similar to using Hotmail for your email, and Company X for your Internet connection – your Internet connection is simply a means of accessing email, or VoIP – no matter which provider you use.
It is entirely up to you whether you cancel your landline and rely solely on voice over IP. Please keep in mind that if you connect to the Internet using DSL technology, you cannot cancel your landline as the connection is required for your Internet connection. What you can do though is select a monthly plan with a low monthly rental, and make all your calls using VoIP (since the plans with low monthly rentals generally have high call costs). If you are on cable, then you do not need to keep your landline.
If you choose to sign up for a full service voice over IP provider you will be assigned a new phone number in the city of your choice. Even if you live in, say, Sydney, and sign up with a provider that gives you a Sydney phone number, you will not be able to port your phone number to your VoIP provider. The reasons are both technical and regulatory, although some Australian providers are working hard towards a solution of local number portability. Essentially it comes down to the fact that your local phone number is linked to a particular Telstra exchange, the collection of phone numbers owned by the VoIP provider, out of which they assign your number, may be attached to a separate Telstra exchange (especially if you are located in a different city). Consequently, due to the geographic nature in which phone numbers are allocated in Australia, there is no way to keep your phone number.
Sending faxes over IP (or FoIP) is very different to voice over IP. One of the premises of voice over IP is that if the sound is changed a little bit by the network, it will not matter, because the human ear at the other end will still be able to hear and understand. This is similar to the principle behind MP3s – music is significantly compressed, but when played back to the human ear, it often sounds indistinguishable from the original. With a voice conversation, if very high frequencies get cut out, or if one part of the sentence arrives a few milliseconds late, or if very quiet sounds aren’t transmitted at all, the receiver is still able to understand perfectly what the other person is saying. Faxes, however, work on a different principle. Like a dial-up modem, they transmit data over the phone line, and in order for the computer (or fax machine) at the other end to be able to interpret what they are saying, the transmission needs to go through unaltered and uncompressed, otherwise it cannot be interpreted properly at the other end. At the very least this means that compression cannot be used for fax over IP. More generally though, faxing over IP is possible, but has to be specifically supported. Many VoIP providers do not support faxing, purely because it adds more overheads to their network, and the benefits of FoIP are not as obvious as VoIP (namely, because there is no compression, and because faxing does not usually involve long phone calls, the savings are not as large).
If you want to get rid of your traditional fax system, you might want to consider an electronic fax service – receive and send faxes via email. There are many such services around, and we can even help you with a trial account to get you started.
Although you can purchase an IP phone or ATA, plug it into your broadband router, and make VoIP calls, this is not the ideal situation. VoIP calls made in this manner are IP-IP calls. In other words, to call someone else, you need to dial their IP address directly. This may work fine in a controlled environment, but has a number of drawbacks:
A much better option is to use a VoIP provider. Your IP phone or ATA can be configured to register with the provider using a unique identification number. Other VoIP users can then call you using this ID number rather than having to know your IP address, as the provider is able to ‘find’ you based on this number. Your IP phone registering with a provider is analagous to your email program on your PC connecting to the server – once connected, the server knows where to deliver your email! Additionally, many providers offer the ability to call the PSTN as well as other VoIP users (this is generally how the providers make their money). By using a provider, you will often receive other features such as voicemail (eg., for when your Internet connection is down) and conference calling.
Note that not all IP phones and ATAs support direct IP-IP dialling. In other words, some IP phones and ATAs will not work at all unless you have them register with a suitable VoIP provider.
The word ‘codec’ is a concatenation of encoer/decoder. With respect to voice over IP, a codec is an algorithm used to encode and potentially compress the analogue voice signal into digital form to be transmitted over the Internet. At the remote end, that same codec is then used to decode and decompress the digital signal in order to reproduce the original analogue signal. There are a variety of different codecs available that offer varying levels of compression and quality. Mostly, the end user does not have to be concerned about codec selection as it is done automatically by your VoIP client and that of the person you are talking to (or the provider if you are calling the PSTN). We have created a summary sheet containing information about some of the more popular codecs used in voice over IP, as well as a table listing codec support amongst the popular VoIP hardware and software.
Calling between SIP based VoIP providers is possible only if the provider you are calling has a policy which allows incoming calls from non-members. Many VoIP providers (but not all) have this policy. To make such a call, you need to dial into your IP phone <phone_number>@othersipprovider.com. This notation is easy to enter when using a softphone as it can be typed in using a keyboard, however, it is not easy from a standard telephone! For this reason, many providers support peering whereby they set up certain prefixes which you can dial to get through to another provider. For example, if you use ATP, you can call a SIPPhone user directly simply by dialling their SIPPhone number (which always begins with 1747 – this is the part that identifies it to ATP as being a SIPPhone number).
Given the number of providers, not all providers peer directly (by having prefixes to dial) with all other providers. This dilema has been solved by SIP Broker – an independent and free service that enables people using different providers to call each other, even if the providers do not support peering prefixes directly.
Online credit card payments are taken through our secure gateway which interfaces directly with the ANZ EPOS Internet Payment Gateway. Your credit card details are not stored on our server at any time, but are passed directly to the bank via a dedicated link to secure verification and processing. Payment can be made using Visa, Mastercard, Bankcard and American Express. For further details, see our policy statement.
Once you have all the equipment, you will be able to make free calls (the only cost is that of having your Internet connection) to other VoIP users using a SIP service provider. A SIP service provider acts as a proxy between you and the person you want to call. Once the call is established, the data is sent between you and the person you called, not through the SIP provider.
Many SIP providers offer the ability to call VoIP users who have registered with a different SIP provider.
Some SIP providers make money through advertising on their website, and by offering customers premium packages which give you the option of making calls from your computer, to normal telephone lines – the prices of which are generally cheaper than traditional telephone companies, even for STD calls in Australia!
Since we target our products and services at New Zealand business’s, all prices mentioned on the our website exclude 15% GST unless stated otherwise.
All prices specified on the Kiwi VoIP website are in NZ dollars unless explicitly stated otherwise.
All personal information collected during the purchase process is used solely to process your order, and provide you with a positive customer experience as outlined in our privacy policy. In short, you can expect to not receive any junk mail (electronic or otherwise) from us or any third party as a result of providing us with your contact details.
Unfortunately we will not match the price of another store if you find the same product sold elsewhere. We try to keep our prices as low as possible as our main philosophy is to get as many consumers using VoIP as possible, though occasionally you may come across a cheaper price elsewhere. The most common source of ‘cheaper prices’ is purchasing from an online store in the USA. There are a number of reasons why some products are more expensive when purchased in NZ – including GST, NZ regulatory compliance, NZ plugs on AC adapters, shipping costs and import taxes. When purchasing from Kiwi VoIP, a fully NZ owned and operated business, you will receive a product that works in NZ and complies with the necessary regulations, and you have access to support at the cost of a local call in the same time-zone as you! Kiwi VoIP is a VoIP specialist and we pride ourselves in this fact. We are not a ‘general’ computer store, but sell only VoIP related products. When you purchase from us you will get the full support you need in setting up, and we are always more than happy to answer questions from any of our past customers. We test out and have used all the hardware we sell and you can be rest assured that your purchase will be satisfying.
Broadly speaking, voice over IP hardware breaks down into two categories; those which are telephones in themselves, and adapters which allow you to use a standard telephone for VoIP. At Kiwi VoIP, most of our products are of the adapter type as they provide the greatest flexibility, allowing you to use any telephone handset you like – including a cordless phone. If you purchase a ‘complete VoIP phone unit’, you have no choice but to use the handset that forms part of the phone.
Within these two categories, one can get fully self contained units, and PC peripheral units. Here are some of the advantages and disadvantages of either type:
Articles on voice over IP are beginning to appear in many computer magaines and even in the mainstream media, Kiwi VoIP has even appeared in a few of them. Additionally, with the growing popularity of VoIP, a lot of consumer oriented sites are springing up. Here is a list of sites we believe contains useful information about Internet telephony – both for new and experienced users.
Not necessarily. To use VoIP you would generally sign up with Kiwi VoIP. Some providers give you the ability to only call other VoIP users, some providers allow you to only call standard telephones, and some allow you to call both. Any time you use a provider that allows you to call standard telephones, you will have to pay, however VoIP-VoIP calls are generally free. The key is to sign up with a provider that suits your needs most, so you do not end up paying for something you do not need (eg., a monthly service fee if you never plan on making calls to the PSTN!)
Every person is different, and so is every situation. Different people use VoIP for different reasons – although the common theme is saving money. We have put together a few typical situations in which VoIP is used. If this does not clear things up for you, have a look at the products and services we offer and see if any of them sound like something you might want.
Many companies are currently replacing their old fashioned PABX’s with newer software based systems. These soft-PABX’s turn an office telephone network into an IP network, and even allow companies to use the same wiring for their computer and telephone network. Companies are starting to save vast amounts of money by doing this since voice conversations are carried over the same network as their data, and this data can be then sent across the Internet to offices across the country, or across the world, for the cost of Internet access. By bypassing the telephone companies for long distance phone calls, companies are saving themselves hundreds of thousands of dollars in inter-office calling. Additionally, by having a phone system based on software, rather than hardware, upgrades can be made quickly and cheaply, new features can be easily implemented, and office workers can even connect to the PABX remotely!
If your business moves towards a complete VoIP solution, whether hosted or in-house, you will still have ‘standard’ incoming phone numbers. They might be the same phone numbers as you currently have, or you might get allocated new incoming phone numbers by the VoIP provider. Either way, for all intents and purposes they are the same as other phone numbers on the telephone network. Therefore, if you have a 08000 service, you can simply point that number to your new incoming phone numbers and all your 0800 calls will come through on your VoIP service.
