So how does VoIP work? VoIP is largely dependent on a protocol called SIP (Session Initiation Protocol). The idea behind SIP is to provide a simple, lightweight means for creating and ending connections for real-time interactive communications over IP networks -- mainly for voice, but also for videoconferencing, chat, gaming or even application sharing. In other words, the protocol initiates call setup, routing, authentication and other communication features to endpoints within an IP domain.

VoIP is quite similar to e-mail. That is, you have the Internet, a server and a client. If users want to check their e-mail they would have to register with their e-mail server and download mail from that server. If they send an e-mail that also goes via that mail server. Nor do users have to be tied to one location -- they can send/receive mail while travelling. As long as they can connect to the Internet, they can use the service.

VoIP is exactly like that. There is a server, in this case a SIP proxy server or a softswitch. These are both the same -- software applications running on general purpose computing. In the first instance, the client (ATA, IP phone or soft phone) will register with a user's SIP server. So, from then on, when a caller picks up the phone to make a phone call, because the ATA has a current registration, there will be dial tone.

This works because the first thing an ATA does when it boots up is get a DHCP address so it has an address on the Internet, and then the DHCP server points to a DNS server on the Internet and it will use that to resolve this unique server and it will register with the service provider to say "hey, I am alive".

As a plug-and-play experience, this whole DHCP event is important. With many ATAs on the market today, users do not need a computer to make it work. All that is required is for the ATA to be powered up from the wall and plugged into a phone.

At the boot-up process, the ATA sends a SIP message (an invite) to the SIP register (softswitch) which is maintained by a user's VoIP service provider. The softswitch says "OK". From then on, there is a two-way communication which happens every 60 seconds, in much the same way as a heartbeat. The softswitch does a lot. It is constantly listening to thousands of people, even if they are not making calls.

VoIP service providers charge by the minute. Ensuring a call is routed efficiently and cheaply also lies with the softswitch.

Once the softswitch determines where the subscriber is, it does a thing called Least Cost Routing (LCR). It works out the least-cost path to get to its endpoint by using algorithms inside the softswitch and by provisioning. Provisioning is the process of entering subscriber information into the softswitch. This is used by the LCR algorithms, so really it's the softswitch's knowledge of the network that allows it to do LCR. For example, if someone dials Europe from a Sydney suburb, it will state the best path to that final destination. All of this is transparent to the end user.

Once the softswitch finds the least cost and sends it on to the gateway (also owned by the VoIP service provider), that gateway will do an IP to TDM (Time Division Multiplexing is the language of the POTS) conversion. It will then send it on to the PSTN via a protocol called SS7, which is the signaling system for the PSTN. In other words, SS7 is the language which telcos use to talk to each other. Once in the PSTN "cloud" it will eventually be routed to a phone number and that phone will then ring.

An analog telephone adapter, or ATA (also referred to as a media terminal adaptor or broadband phone adaptor), allows you to enable an ordinary telephone, including your cordless, to make VoIP calls. All you need to do is plug your analog phone into the ATA, then take your cable from the Internet port of the ATA and plug that into a spare Ethernet port on your ADSL or cable router. If you don't have a spare port you will need a switch to provide a spare Ethernet port.

Current ATA devices will set you back between $145 and $210, depending on their capabilities.

An ATA may sound like a modem, but in fact there is quite a bit of difference in the hardware technology. An ATA is specifically designed to deal with the voice traffic and not data traffic. It takes the analog human voice and converts that to an IP packet with the use of the compression codec, then sends that over the Internet.

There are several features to look out for in ATAs:

Codec: VoIP devices, including the ATA, use codecs to compress outgoing and decompress incoming voice. The codecs in an ATA receive the data and break it into small files, or packets, for sending and receiving across the IP network. There are hundreds of codecs which can be used for VoIP and as a result the quality levels differ with each one used.

The G.711 is the codec that the PSTN (Public Switched Telephone Network) uses. It is quite large for travel across the Internet, enabling 64Kbps of voice. G.729, for example, compresses voice down to 8Kbps or 20Kbps when the packet header is factored in. This packet works out to be a quarter the size, yet still maintains a high quality, and is therefore ideal for sending over an IP network. G.729 is the most commonly deployed codec in Australia.

(See table at bottom of page).

CPU performance: An ATA might say it supports a bunch of codecs (G.711, G.729 or G.722). But if one codec is more CPU intensive than the other, it needs more MIPS (millions of instructions per second) to run it. Therefore a more powerful ATA will provide better voice quality as it compresses and processes that codec a lot better than a modem with less power.

Connectivity: USB or Ethernet. Ethernet is best because it gives users the versatility of plugging in multiple devices. Whereas, if the connection was a USB ATA it would need to go via a PC.

IP Phones look pretty much the same as normal analog phones. The main difference here is that they are digital, and have an inbuilt ATA. Additionally, rather than having the standard RJ-11 phone connectors, IP phones use an RJ-45.

An IP phone is ideal for businesses, where Ethernet network ports are distributed around the office space. All that is required is for the user to plug the phone straight into an Ethernet port and away they go.

Assuming users already have broadband, the type of modem they have is important.

A home or office modem should be Network Address Translation (NAT)-enabled. NAT is a protocol used by a device such as a firewall, router or computer, and provides a mapping between internal IP addresses and public IP addresses in real time. The modem may be NAT-enabled, with a DHCP server in it (Dynamic Host Configuration Protocol is for assigning dynamic IP addresses to devices on a network). In most cases, when the ATA is plugged in, it will boot up and automatically seek out the softswitch of the service provider to which it is connected. More on softswitches later, but for now it is important to note that the softswitch is owned by the VoIP service provider and will direct phone calls to where they are supposed to be going.

If users have an old broadband account, it is likely that their modem may not be up to scratch for voice delivery.

Users must look for these requirements in a modem:

It must terminate L2TP or PPTP: Some ISPs have a point-to-point session used for authentication. If a point-to-point session goes from user's ISP to the software on their PC, they are in trouble. This is because as soon as they plug in their ATA, their ISP is going to see the packets coming from their ATA and say "you are not authenticated". If the modem authenticates, then everything behind it will get through. Some of the older and cheaper modems don't do L2TP because of additional cost.

Ethernet on the modem: USB is inferior to Ethernet because it means the modem needs to be connected to the PC to talk. In this case, users would be limited to a soft phone.

CPU power: Modems are CPU based. A slow CPU means it can only do so many packets per second. A VoIP phone call transmits at 100 packets per second (50 packets per second each way). That is the sort of performance that every phone call is going to require of a modem (and ATA). Therefore a modem must be able to handle that traffic.

Soft phones, as their name suggests, are not hardware devices. The purely software services, such as FreshTel, Skype or BroadBand Phone require users to install small software applications, or clients, on their PC to run the service. In this case, users need to connect a USB/FireWire handset or microphone headset to a PC to talk.

The key point to note here is that users need to have their PC turned on at all times to make and receive calls.

The broadband connection is quite important. The minimum required for VoIP is ideally 256Kbps download. Users can get away with 128/64Kbps, but the bigger the pipes at home the better the VoIP service as it reduces the chance of voice traffic having to compete with other (larger) data files such as MP3 or images. Typically, a VoIP call will consume about 20Kbps of bandwidth. So if a user has a 128/64Kbps service, the call will use about 15 per cent download and about 30 per cent of the upload bandwidth available.

But speed is not the only thing that matters. The actual broadband service is also critically important. Ultimately, a VoIP service is totally dependent on a reliable broadband service provider to help deliver voice traffic. If a user's ADSL/cable/wireless broadband service provider has continual dropouts or outages, it will obviously affect the VoIP service. You cannot surf the Internet and open e-mails when there are outages. Nor can you make VoIP calls.

It is worth noting that although users are doing away with their telephone provider if they chose to use a VoIP service, they are still required to pay the telephone line rental for the ADSL-enabled phone line. This is because ADSL is an add-on service for telephone subscribers. If users are on cable, they would be subject to the cable provider's subscription rates.

A service provider offers the connection to other VoIP services or to the PSTN. In other words, they are responsible for directing your call to where it is supposed to go.

As with any other type of service provider, VoIP service levels will differ. Most service providers charge a monthly fee, then additional costs when calls are made. Service providers offer different rates based on their services. Charges to look out for are local call rates, STD rates, mobile rates, and rates to any international destinations that may be of interest to the consumer. These prices are charged by the minute and are continually negotiated by the service provider, other VoIP providers and telcos around the world. As a result, call prices can change rapidly, so it is important to keep an eye on the cost of your calls.

Users should note that there are differences between public and private numbering schemes.

If the user only wants to make cheap outgoing calls, and receive calls within the service provider's own network, then a private number is sufficient.

Some services also enable users to receive calls from regular telephones -- at no extra cost. This feature is known as DID (Direct Inward Dialling) and, although you can't use your existing number for the moment, you can get a real, regular eight-digit format number for people to call.