IP Telephony, also called 'Internet telephony', is the technology that makes it possible to have a telephone conversation over the Internet or a dedicated Internet Protocol (IP) network instead of dedicated voice transmission lines. This allows the elimination of circuit switching and the associated waste of bandwidth. Instead, packet switching is used, where IP packets with voice data are sent over the network only when data needs to be sent, i.e., when a caller is talking.
Its advantages over traditional telephony include:
- lower costs per call, or even free calls, especially for long-distance calls.
- lower infrastructure costs: once IP infrastructure is installed, no or little additional telephony infrastructure is needed.
- new advanced features.
- A higher degree of reliability and resilience
- future proof as functionality is software (protocol) based and does not require hardware replacement
Voice over IP traffic does not necessarily have to travel over the global Internet; it may also be deployed on private IP networks for example on a LAN inside a single building.
The protocols used to carry the signal over the IP network are commonly referred to as Voice over IP or VoIP protocols.
Corporate and telco use of VoIP
Although few office environments and even fewer homes use a pure VoIP infrastructure, telecommunications providers routinely use IP telephony, often over a dedicated IP network, to connect between their switching stations, where they convert the dedicated voice signal to IP packets and back. The result is a data-abstracted digital network which the provider can easily upgrade and use for multiple purposes. Corporate customer support centers which provide support over telephone often use IP telephony exclusively to take advantage of the data abstraction that comes with it. The benefit of using this technology is the need for only one class of circuit connection and better use of the available bandwidth. IP telephony is commonly used to route traffic that may be originated from and terminated at conventional PSTN (Public Switched Telephone Network) telephones. VoIP is now widely deployed by carriers, especially for international telephone calls. Users are typically completely unaware that their telephone call is being routed over IP infrastructure for most of its distance instead of the circuit switched PSTN. VoIP is also used by large companies to eliminate call charges between their offices, by using their data network to carry inter-office calls. They may also use VoIP to reduce the costs of calls outside the company by carrying them to the nearest point on their network before handing them off to the PSTN. There are companies which offer a gateway to the PSTN from any VoIP phone. You can simply dial a conventional telephone number and the telephone call will be routed over your internet connection to the company that operates the gateway, and they will bill you, not the local phone company. Electronic Numbering (ENUM) makes it possible to dial traditional E.164 phone numbers, but be connected entirely over the Internet if the other party uses Enum, so you do not incur any expenses aside from the Internet connection fee. It's also possible for companies to buy their own gateway to eliminate costs of a third party, which can be cost effective in some situations.
VoIP implementation challenges
Because IP does not by default provide any mechanism to ensure that data packets are delivered in sequential order, or provide any Quality of Service guarantees, implementations of VoIP face problems dealing with latency and possible data integrity problems.
One of the central challenges for VoIP implementers is restructuring streams of received IP packets, which can come in any order and have packets missing, to ensure that the ensuing audio stream maintains a proper time consistency. Another important challenge is keeping packet latency down to acceptable levels on satellite circuits.
- The network operator can also ensure that there is enough bandwidth end-to-end to guarantee low-latency, high quality voice. This is easy to do in private networks, but much harder to do at less than 256 kbit/s without a fragmentation mechanism.
In the overwhelming majority of implementations, RTP is used to transmit VoIP traffic ("media"). Notable exception is IAX which carries both signaling and voice data over a single UDP stream, which results in fewer problems while traversing firewall and NAT devices.
For signaling, there are several alternative protocols:
- SIP, the IETF Session Initiation Protocol, a newcomer gaining popularity
- H.323, the ITU's widely deployed and continually updated VoIP protocol carrying billions of minutes of traffic each month
- Skinny Client Control Protocol, proprietary protocol from Cisco
- Megaco (a.k.a. H.248) and MGCP, both media gateway control protocols
- MiNET, proprietary protocol from Mitel
- IAX, the Inter-Asterisk eXchange protocol used by the Asterisk open source PBX server and associated client software
Mass-market telephony over broadband Internet access
A new development has been the introduction of mass-market VoIP services over broadband Internet access services, in which subscribers make and receive calls as they would over the PSTN. This requires an analog telephone adapter (ATA) to connect a telephone to the broadband internet connection. Companies in the US, such as Verizon, Vonage, VoicePulse, and Packet8, use IP to offer unlimited calling to the US, and sometimes to Canada or to selected countries in Europe or Asia, for a flat monthly fee. One advantage of this is the ability to make and receive calls as you would at home, anywhere in the world, at no extra cost. As calls go via IP, this does not incur charges as call diversion does via the PSTN, and the called party does not have to pay for the call.
For example, somebody may call you on a number with a US area code, but you could be in London, and if you were to call another number with that area code, it would be treated as a local call, regardless of where you are in the world. However, the broadband phone is likely to complement, rather than replace a PSTN line, as it still needs a power supply, while calling the US emergency services number 911, may not automatically be routed to the nearest local emergency dispatch center, or be of any use for subscribers outside the US.
Another challenge for these services is the proper handling of outgoing calls from Fax machines, TiVo/ReplayTV boxes, satellite television receivers, alarm systems, conventional modems or FAXmodems, and other similar devices that depend on access to a voice-grade telephone line for some or all of their functionality. At present, these types of calls sometimes go through without a hitch, but in other cases they won't go through at all. And in some cases, this equipment can be made to work over a VoIP connection if the sending speed can be changed to a lower bits per second rate. If VoIP and cellular substitution becomes very popular, some ancillary equipment makers may be forced to redesign equipment, because it would no longer be possible to assume a conventional voice-grade telephone line would be available in almost all homes in North America and Western-Europe. The TestYourVoIP website offers a free service to test the quality of or diagnose an internet connection by placing simulated VoIP calls from any Java-enabled web browser, or from any phone or VoIP device capable of calling the PSTN network.
There is also a free service called Free World Dialup (FWD), that permits users to make free telephone calls to other FWD users, although has only limited connections to and from the public switched telephone network.