A guide to VoIP telephony

What is VoIP, how much does it cost, how does it work and what are the alternatives? Our Tech Guide provides the answers.

VoIP, or Voice over Internet Protocol, is a method of sending audio voice signals -- or, more specifically, telephone signals -- as data packets over the Internet. By taking advantage of the different charging structure of broadband services, the cost of telephone calls can be drastically reduced.

The legacy of the analogue telephone system
The early telephony systems were, of course, entirely analogue and used a network of copper wires to carry the signal current over a one-to-one connection between two telephones. The legacy of this system creates a huge degree of physical inertia to change: although all major telephone backbones between exchanges are now digital and carried by fibre optic, microwave and satellite links, the local loops between the exchanges and the home or business still use the many miles of existing ‘analogue’ copper wiring.

Broadband technology exploits the maximum possible capacity of the old wiring to run digital signals over these copper connections. Of course, broadband data links aren’t limited to these copper wire connections, but are also available over cable or, in some cases, satellite links.

The old PSTN (Public Switched Telephone Network) telephony system of point-to-point connections provided by a monopoly service provider (British Telecom in the UK) supports a line rental plus a charge-per-call payment model, with rates set by the monopoly provider at what it feels the market will bear. By contrast, broadband is an ‘always-on’ system with multiple competing service providers that connects to the Internet, where data packets travel by diverse routes: this means that the charging structure is quite different, and VoIP calls generally cost less.

VoIP call charges
Often touted as a ‘free calls’ telephone system, in reality VoIP calls can be divided into three types, which attract different charge rates. These are: calls between two VoIP phones; outgoing calls from a VoIP phone to a PSTN phone; and incoming calls from a PSTN phone to a VoIP phone.

Calls made between two VoIP subscribers using the same service provider are treated as normal Internet traffic and costs are covered by the users’ broadband connection fees. These are often regarded as ‘free’ calls.

Calls to PSTN numbers are charged for by the VoIP service provider since they must pass through a VoIP-to-PSTN gateway maintained by the service provider.

Incoming PSTN calls to a VoIP phone are charged to the caller at a rate that depends on the type of conventional number assigned to the VoIP phone. National rate numbers are usually assigned by the VoIP service provider at no cost to the VoIP subscriber. The VoIP service provider then pockets some of the call charges paid by incoming calls. Incoming calls to these numbers cost the caller around 4 pence per minute. Alternatively, the VoIP subscriber can opt to pay a fee for a local code number, in which case incoming calls are charged to the caller at local rates.

Dialling a number with a VoIP phone
The complexity of actually placing a call with a VoIP phone varies, depending on the nature of the required connection. Calls made between VoIP phones using the same service provider can be placed by simply dialling the service provider ID number for the destination phone.

Calls between service providers can be made by dialling the destination ID plus the service provider domain. Inter-service connections are limited at the moment, with each provider only supporting calls to a select number of other services. Inter-domain calls become a bit tricky if you're using only the phone keypad to enter the text in the domain name.

Calls to PSTN numbers can be dialled just like making a normal phone call, with the caveat that the number must always include the full area code.

In theory it’s also possible to dial another VoIP phone using its IP address. However, since most VoIP phones will be connected behind a firewall via a gateway that's using NAT (Network Address Translation), this usually isn’t practical.

How VoIP works
VoIP works by converting speech into a digital signal, compressing it and encoding it using one of a range of speech CODECs (COder/DECoder) and breaking it into data packets that are sent over the Internet. All VoIP systems support a range of CODECs that can juggle call quality and bandwidth. Call quality, or QoS (Quality of Service), is affected by Internet traffic density and, on any individual call, the call may switch to a lower bandwidth (and therefore lower quality) CODEC if the available Internet bandwidth is limited.

At the destination, the packets are reassembled, decoded using the appropriate CODEC, and converted to analogue audio that can be heard by the call recipient. The compression introduced by the CODEC and the packet switching makes VoIP much more efficient, and therefore cheaper, than a point-to-point analogue phone call. The standard Internet protocols are used to route the packets between addresses just like any other internet traffic -- hence the term Voice over Internet Protocol. Additional protocols are required to handle call control.

Unlike the traditional telephone service, the destination ID (or telephone number) is not assigned to a specific run of cable, but by the user to the VoIP device itself. This means that VoIP IDs/phone numbers are portable. A VoIP phone can be moved by the user to various locations and it will still receive calls made to its ID.

All VoIP devices (apart from software VoIP systems like Skype) have a built-in Web server. Each VoIP device must be assigned an IP address. If the device is a phone, this can be done either manually from the phone's keypad or via a DHCP server. Users can then use a browser to gain protected access to this Web server by entering a user name and password. Once access is gained, the user can set the VoIP ID, password, domain and an outbound proxy. These values are usually stored in NVRAM inside the VoIP device.

Setting up a VoIP device can be tricky because, as with so much network equipment, terms and syntax vary widely from one manufacturer to another. The VoIP service providers are the best source of setup information, and each provider supports a limited range of phones, so your choice of hardware and provider are quite closely linked.

VoIP protocols -- SIP & H.323
Handling telephone calls over the Internet requires an extra protocol, in addition to the IP, to manage call setup, voice data transfer and call termination. There are several standard protocols designed to do this; ITU-T H.323 (known as H.323 for short), Media Gateway Control Protocol (MGCP) and the IETF Session Initiation Protocol otherwise known as SIP. The latter is one of the simplest and by far the most popular: SIP is used by Vonage and many other VoIP service providers.

There are also proprietary VoIP protocols -- the Minet protocol that Mitel Networks uses on its range of VoIP equipment and the peer-to-peer Skype protocol, for example.

What’s required to use VoIP?
To use VoIP it’s necessary to have a VoIP service provider -- just as an ISP (or Internet Service Provider) is required to use the Internet. You also, of course, need a broadband connection. Some variety of hardware to perform the phone audio functions is required. PC-integrated services require VoIP client software, while standalone VoIP phones can be set up with the service information from their keypads, although it's usually easier to use a Web browser, telnet or FTP software to enter these details.

VoIP hardware and software
The simplest VoIP systems make use of standard computer sound cards, fitted with a microphone and loudspeakers (or a headset), to stand in for the traditional telephone handset. Another option is to use a USB VoIP handset that's designed to resemble a normal telephone. VoIP client software supplied by the VoIP service provider handles a telephone book, dialling and answering calls. The drawback of these integrated systems is that because they rely on a PC, the PC must be turned on to make or receive calls. The advantage is that they are inexpensive, often easy to set up and the software phone book makes it very easy to organise and dial contacts.

A discrete solution in the form of either a VoIP phone, or a conventional phone connected to an ATA (Analogue Telephone Adapter), is often more practical.


A selection of VoIP phones, from Grandstream and Linksys.


VoIP phones look and operate much like conventional phones except that they are designed to plug in to an Ethernet network. They don’t require the presence of a computer on the network, other than for initial setup (although they can be set up with some difficulty using only the keypad). It's possible to plug a configured VoIP phone directly into a broadband gateway router -- or with a fixed IP service, directly into a broadband modem. A basic VoIP phone like the Grandstream BudgeTone 101 costs £47.98 (inc. VAT).


ATAs like these from Linksys and Grandstream can be used to convert a conventional telephone for VoIP use.


ATAs can be used to convert ordinary analogue phones for VoIP use. These adapters connect to the phone and to an Ethernet network and have built-in Web servers. An ATA like the Linksys SPA1001 Phone Adapter costs £35.25 (inc. VAT). Recently some broadband gateway routers have appeared with built-in adapters for a number of conventional phones.

VoIP service providers
VoIP service providers come in a variety of flavours, supporting either proprietary or open standards like SIP. Each provider has a slightly different charging structure and may be oriented to different market sectors. For example, at the entry level, Skype, Callserve and Tesco use proprietary protocols and are aimed at the home user. More advanced services like Demon, Sipgate, Sipphone, Voipfone, Voiptalk and Vonage all use SIP.

A typical VoIP-to-VoIP connection


This diagram shows the elements involved in a typical VoIP-to-VoIP connection.
 
 

Alternatives to VoIP
VoIP isn’t the only game in town when it comes to Internet telephony: there’s also TDMoIP or Time Division Multiplexing over Internet Protocol. Most companies with more than two or three employees use a PBX or PABX (Private Branch eXchange or Private Automatic Branch eXchange) to handle their internal telephone system. Small PBXs can operate with multiple ordinary phone lines, but once the employee count rises and a company physically spreads out over a wide area it’s usual to switch to a Time Division Multiplexed PBX connected to one or more wideband lines. For example a T1 line has a bandwidth of 1.544Mbps and can handle up to 24 extensions and a T2 line has a bandwidth of 43.232Mbps and can handle up to 672 extensions.

Wideband lines are used to transfer the private exchange signals between localised branches -- for example, between a head office and a factory. These wideband lines are leased from the telephone company and are relatively expensive.

TDM samples the audio signals of each call for a fixed length of time. These samples are then strung together in sequence (multiplexed) for transfer via the wideband line.

The installation of a TDM PBX often represents a considerable infrastructure investment, which companies may not wish to write off by changing over to a VoIP-based system. However, it's still possible to save money on leased line costs by installing one, or more, links using TDMoIP gateways. A TDMoIP gateway is a piece of hardware that accepts the multiplexed feed out of a TDM PBX and has a broadband connection to the Internet. On the PBX side, these TDMoIP gateways can emulate all the varieties of wideband line normally used for multiplexed telecoms, from T1/E1 to T3/E3. This is often called ‘pseudowire emulation’ by the TDMoIP vendors.

Running TDM, a system that depends on accurate and dependable timing, over the Internet, which is inherently asynchronous, requires techniques that are too complex to go into here. TDMoIP gateway products are differentiated by how well they handle these timing problems. TDMoIP gateways can be purchased from Afar Communications or RAD Data Communications.

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