Voice-over-Internet protocol (VoIP, IPA: /vɔjp/) is a protocol optimized for the transmission of voice through the Internet or other packet-switched networks. VoIP is often used abstractly to refer to the actual transmission of voice (rather than the protocol implementing it). This latter concept is also referred to as IP telephony, Internet telephony, voice over broadband, broadband telephony, and broadband phone.
VoIP providers may be viewed as commercial realizations of the experimental Network Voice Protocol (1973) invented for the ARPANET providers. Some cost savings are due to utilizing a single network to carry voice and data, especially where users have underused network capacity that can carry VoIP at no additional cost. VoIP-to-VoIP phone calls are sometimes free, while VoIP calls connecting to public switched telephone networks (VoIP-to-PSTN) may have a cost that is borne by the VoIP user.
Voice-over-IP systems carry telephony signals as digital audio, typically reduced in data rate using speech data compression techniques, encapsulated in a data-packet stream over IP.
There are two types of PSTN-to-VoIP services: Direct inward dialing (DID) and access numbers. DID will connect a caller directly to the VoIP user, while access numbers require the caller to provide an extension number for the called VoIP user.
Voice-over-Internet Protocol has been a subject of interest almost since the first computer network. By 1973, voice was being transmitted over the early Internet.[1] The technology for transmitting voice conversations over the Internet has been available to end-users since at least the early 1980s. In 1996, a shrink-wrapped software product called VocalTec Internet Phone (release 4) provided VoIP along with extra features such as voice mail and caller ID. However, it did not offer a gateway to the PSTN, so it was only possible to speak to other Vocaltec Internet Phone users.[2] In 1997, Level 3 began development of its first softswitch (a term they invented in 1998); softswitches were designed to replace traditional hardware telephone switches by serving as gateways between telephone networks.[3]
Revenue in the total VOIP industry in the US is set to grow by 24.3% in 2008 to $3.19 billion. Subscriber growth will drive revenue in the VOIP sector, with numbers expected to rise by 21.2% in 2008 to 16.6 million. The US's largest VOIP provider is Vonage[4].
[edit] Functionality
VoIP can facilitate tasks and provide services that may be more difficult to implement or more expensive using the PSTN. Examples include:
* The ability to transmit more than one telephone call over the same broadband connection. This can make VoIP a simple way to add an extra telephone line to a home or office.
* Conference calling, call forwarding, automatic redial, and caller ID; zero- or near-zero-cost features that traditional telecommunication companies (telcos) normally charge extra for.
* Secure calls using standardized protocols (such as Secure Real-time Transport Protocol.) Most of the difficulties of creating a secure phone connection over traditional phone lines, like digitizing and digital transmission, are already in place with VoIP. It is only necessary to encrypt and authenticate the existing data stream.
* Location independence. Only an Internet connection is needed to get a connection to a VoIP provider. For instance, call center agents using VoIP phones can work from anywhere with a sufficiently fast and stable Internet connection.
* Integration with other services available over the Internet, including video conversation, message or data file exchange in parallel with the conversation, audio conferencing, managing address books, and passing information about whether others (e.g. friends or colleagues) are available to interested parties.
* Advanced Telephony features such as call routing, screen pops, and IVR implementations are easier and cheaper to implement and integrate. The fact that the phone call is on the same data network as a users PC opens a new door to possibilities.
[edit] Implementation
Because UDP does not provide a mechanism to ensure that data packets are delivered in sequential order, or provide Quality of Service (QoS) guarantees, VoIP implementations face problems dealing with latency and jitter. This is especially true when satellite circuits are involved, due to long round-trip propagation delay (400–600 milliseconds for links through geostationary satellites). The receiving node must restructure IP packets that may be out of order, delayed or missing, while ensuring that the audio stream maintains a proper time consistency. This function is usually accomplished by means of a jitter buffer in the voice engine.
Another challenge is routing VoIP traffic through firewalls and address translators. Private Session Border Controllers are used along with firewalls to enable VoIP calls to and from protected networks. Skype uses a proprietary protocol to route calls through other Skype peers on the network, allowing it to traverse symmetric NATs and firewalls. Other methods to traverse firewalls involve using protocols such as STUN or ICE.
VoIP challenges:
* Available bandwidth
* Network Latency
* Packet loss
* Jitter
* Echo
* Security
* Reliability
* In rare cases, decoding of pulse dialing
Many VoIP proiders do not decode pulse dialing from older phones. The VoIP user may use a pulse-to-tone converter, if needed.[citation needed]
Fixed delays cannot be controlled but some delays can be minimized by marking voice packets as being delay-sensitive (see, for example, Diffserv).
The principal cause of packet loss is congestion, which can sometimes be managed or avoided. Carrier VoIP networks avoid congestion by means of teletraffic engineering.
Variation in delay is called jitter. The effects of jitter can be mitigated by storing voice packets in a jitter buffer upon arrival and before producing audio, although this increases delay. This avoids a condition known as buffer underrun, in which the voice engine is missing audio since the next voice packet has not yet arrived.
Common causes of echo include impedance mismatches in analog circuitry and acoustic coupling of the transmit and receive signal at the receiving end.
