Voice-over-Internet protocol (VoIP, ) 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). VoIP allows users to use regular telephone networks anywhere through any internet service provider, and avoids issues on long distance charges that are normally subject to callers. 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. VoIP can be a great benefit to major corporations trying to cut costs by simply running network cables. Although it requires those using VoIP to have a well configured network it is a new use of IP that is extremely cost efficient. Skype and Vonage are great examples of how VoIP is being greatly utilized world wide.
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. 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. 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.
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 United States' largest VoIP provider is Vonage.
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.
Many VoIP providers do not decode pulse dialing from older phones. The VoIP user may use a pulse-to-tone converter, if 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.
Voice travels over the internet in almost the same manner as data does in packets. So when you talk over an IP network your conversation is broken up into small packets. These voice and data packets travel over the same network with a fixed bandwidth. This system is more prone to congestion and DoS attacks than traditional circuit switched systems.
To increase the reliability of VoIP phones, the VoIP provider needs to increase dedicated and redundant connectivity via T1 access and backup DSL, with automatic failover at each location. The company can create a reliable network by reducing the number of single points of failure and providing its own UPS or other backup power generators on site.
It has been suggested to rely on the packetized nature of media in VoIP communications and transmit the stream of packets from the source phone to the destination phone simultaneously across different routes (multi-path routing). In such a way, temporary failures have less impact on the communication quality. In capillary routing it has been suggested to use at the packet level Fountain codes or particularly raptor codes for transmitting extra redundant packets making the communication more reliable.
A number of protocols have been defined to support the reporting of QoS/QoE for VoIP calls. These include RTCP XR (RFC3611), SIP RTCP Summary Reports, H.460.9 Annex B (for H.323), H.248.30 and MGCP extensions. The RFC3611 VoIP Metrics block is generated by an IP phone or gateway during a live call and contains information on packet loss rate, packet discard rate (due to jitter), packet loss/discard burst metrics (burst length/density, gap length/density), network delay, end system delay, signal / noise / echo level, MOS scores and R factors and configuration information related to the jitter buffer.
RFC3611 VoIP metrics reports are exchanged between IP endpoints on an occasional basis during a call, and an end of call message sent via SIP RTCP Summary Report or one of the other signaling protocol extensions. RFC3611 VoIP metrics reports are intended to support real time feedback related to QoS problems, the exchange of information between the endpoints for improved call quality calculation and a variety of other applications.
VoIP is clearly identified as a Least Cost Routing (LCR) system, which is based on checking the destination of each telephone call as it is made, and then sending the call via the network that will cost the customer the least. With GSM number portability now in place, LCR providers can no longer rely on using the network root prefix to determine how to route a call. Instead, they now need to know the actual network of every number before routing the call.
Therefore, VoIP solutions also need to handle MNP when routing a voice call. In countries without a central database like the UK it might be necessary to query the GSM network about the home network a mobile phone number belongs to. As VoIP starts to take off in the enterprise markets because of least cost routing options, it needs to provide a certain level of reliability when handling calls.
MNP checks are important to assure that this quality of service is met; by handling MNP lookups before routing a call and assuring that the voice call will actually work, VoIP companies give businesses the necessary reliability they look for in an internet telephony provider. UK-based messaging operator Tyntec provides a Voice Network Query service, which helps not only traditional voice carriers but also VoIP providers to query the GSM network to find out the home network of a ported number.
In countries such as Singapore, the most recent Mobile number portability solution is expected to open the doors to new business opportunities for non-traditional telecommunication service providers like wireless broadband providers and voice over IP (VoIP) providers.
In November 2007, the Federal Communications Commission in the United States released an order extending number portability obligations to interconnected VoIP providers and carriers that support VoIP providers.
The T.38 protocol is designed to work like a traditional fax machine and can work using several configurations. The fax machine could be a traditional fax machine connected to the PSTN, or an ATA box (or similar). It could be a fax machine with an RJ-45 connector plugged straight into an IP network, or it could be a computer pretending to be a fax machine. Originally, T.38 was designed to use UDP and TCP transmission methods across an IP network. The main difference between using UDP and TCP methods for a FAX is the real time streaming attributes. TCP is better suited for use between two IP devices. However, older fax machines, connected to an analog system, does benefit from UDP near real-time characteristics.
There have been updated versions of the T.30 to resolve the FoIP issues, which is the core fax protocol. Some new fax machines have T.38 built-in capabilities which allow the user to plug right into the network with minimal configuration changes. A unique feature of T.38 is that each packet contains a copy of the main data in the previous packet. This is an option and most implementations seem to support it. This forward error correction scheme makes T.38 far more tolerant of dropped packets than using VoIP. With T.38, it requires two successive lost packets to actually lose any data. The data you lose will only be a small piece, but with the right settings and error correction mode, there is a high probability that you will receive the whole transmission.
Tweaking the settings on the T.30 and T.38 protocols could also turn your unreliable fax into a robust machine. Some fax machines pause at the end of a line to allow the paper feed to catch up. This is good news for packets that were lost or delayed because it gives them a chance to catch up. However, if this were to happen on every line, your fax transmittal would take a long time. Another possible solution to overcome the drawback is to treat the fax system as a message switching system, which does not need a real-time data transmission (such as sending a fax as an email attachment (see Fax) or remote printout (see Internet Printing Protocol)). The end system can completely buffer the incoming fax data before displaying or printing the fax image.
VoIP Enhanced 911 (E911) is another method by which VoIP providers in the United States are able to support emergency services. The VoIP E911 emergency-calling system associates a physical address with the calling party's telephone number as required by the Wireless Communications and Public Safety Act of 1999. All "interconnected" VoIP providers (those that provide access to the PSTN system) are required to have E911 available to their customers. VoIP E911 service generally adds an additional monthly fee to the subscriber's service per line, similar to analog phone service. Participation in E911 is not required and customers can opt-out or disable E911 service on their VoIP lines, if desired. VoIP E911 has been successfully used by many VoIP providers to provide physical address information to emergency service operators.
One shortcoming of VoIP E911 is that the emergency system is based on a static table lookup. Unlike in cellular phones, where the location of an E911 call can be traced using Assisted GPS or other methods, the VoIP E911 information is only accurate so long as subscribers are diligent in keeping their emergency address information up-to-date. In the United States, the Wireless Communications and Public Safety Act of 1999 leaves the burden of responsibility upon the subscribers and not the service providers to keep their emergency information up to date.
A tragic example of a miscommunication with VoIP is the death of 18-month-old Elijah Luck in Calgary, Canada. In an emergency, 911 services were called. An ambulance was sent to the former home of the Lucks. The VoIP telephone company knew the correct address, as they were paying their bill from the correct current billing address the company had on record. "It's up to subscribers to ensure the company has up-to-date contact information" was the response from the VoIP company. After about a half hour wait, the Lucks called from a neighbor's land line, whereupon emergency services arrived in six minutes. Elijah Luck was pronounced dead at the Alberta Children's Hospital.
Phones like the NEC N900iL, and later many of the Nokia Eseries and several WiFi enabled mobile phones have SIP clients hardcoded into the firmware. Such clients operate independently of the mobile phone network unless a network operator decides to remove the client in the firmware of a heavily branded handset. Some operators such as Vodafone actively try to block VoIP traffic from their network and therefore most VoIP calls from such devices are done over WiFi.
Several WiFi only IP hardphones exist, most of them supporting either Skype or the SIP protocol. These phones are intended as a replacement for PSTN based cordless phones but can be used anywhere where WiFi internet access is available.
Another addition to hand held devices are ruggedized bar code type devices that are used in warehouses and retail environments. These type of devices rely on "inside the 4 walls" type of VoIP services that do not connect to the outside world and are solely to be used from employee to employee communications.
In a few cases, VoIP providers may allow a caller to spoof the caller ID information, potentially making calls appear as though they are from a number that does not belong to the caller. Business grade VoIP equipment and software often makes it easy to modify caller ID information. Although this can provide many businesses great flexibility, it is also open to abuse.
These services take a wide variety of forms which can be more or less similar to traditional POTS. At one extreme, an analog telephone adapter (ATA) may be connected to the broadband Internet connection and an existing telephone jack in order to provide service nearly indistinguishable from POTS on all the other jacks in the residence. This type of service, which is fixed to one location, is generally offered by broadband Internet providers such as cable companies and telephone companies as a cheaper flat-rate traditional phone service. Often the phrase "VoIP" is not used in selling these services, but instead the industry has marketed the phrases "Internet Phone", "Digital Phone" or "Softphone" which is aimed at typical phone users who are not necessarily tech-savvy. Typically, the provider touts the advantage of being able to keep one's existing phone number.
At the other extreme are services like Gizmo Project and Skype which rely on a software client on the computer in order to place a call over the network, where one user ID can be used on many different computers or in different locations on a laptop. In the middle lie services which also provide a telephone adapter for connecting to the broadband connection similar to the services offered by broadband providers (and in some cases also allow direct connections of SIP phones) but which are aimed at a more tech-savvy user and allow portability from location to location. One advantage of these two types of services is the ability to make and receive calls as one would at home, anywhere in the world, at no extra cost. No additional charges are incurred, as call diversion via the PSTN would, and the called party does not have to pay for the call. For example, if a subscriber with a home phone number in the U.S. or Canada calls someone else within his local calling area, it will be treated as a local call regardless of where that person is in the world. Often the user may elect to use someone else's area code as his own to minimize phone costs to a frequently called long-distance number.
For some users, the broadband phone complements, rather than replaces, a PSTN line, due to a number of inconveniences compared to traditional services. VoIP requires a broadband Internet connection and, if a telephone adapter is used, a power adapter is usually needed. In the case of a power failure, VoIP services will generally not function without a backup power supply like a common UPS system used in most office buildings. Additionally, a call to an emergency services number may not automatically be routed to the nearest local emergency dispatch center. Some VoIP providers only handle emergency call for one country. Some VoIP providers offer users the ability to register their address so that emergency services work as expected.
Another challenge for these services is the proper handling of outgoing calls from fax machines, DVR 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 any problems, but in other cases they will not 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 Web site 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.
Because of the bandwidth efficiency and low costs that VoIP technology provides, businesses are slowly beginning to migrate from traditional copper-wire telephone systems to VoIP systems to reduce their monthly phone costs.
Corporate customer telephone support often use IP telephony exclusively to take advantage of the data abstraction. The benefit of using this technology is the need for only one class of circuit connection and better bandwidth use. Companies can acquire their own gateways to eliminate third-party costs, which is worthwhile in some situations.
VoIP is widely employed by carriers, especially for international telephone calls. It is commonly used to route traffic starting and ending at conventional PSTN telephones.
Many telecommunications companies are looking at the IP Multimedia Subsystem (IMS) which will merge Internet technologies with the mobile world, using a pure VoIP infrastructure. It will enable them to upgrade their existing systems while embracing Internet technologies such as the Web, email, instant messaging, presence, and video conferencing. It will also allow existing VoIP systems to interface with the conventional PSTN and [mobile phone]s.
Electronic Numbering (ENUM) uses standard phone numbers (E.164), but allows connections entirely over the Internet. If the other party uses ENUM, the only expense is the Internet connection. Virtual PBX (or IP PBX) allow companies to control their internal phone network over an existing LAN and server without needing to wire a separate telephone network. Users within this environment can then use standard telephones coupled with an FXS, IP Phones connected to a data port or a Softphone on their PC. Internal VoIP phone networks allow outbound and inbound calling on standard PSTN lines through the use of FXO adapters.
Ham Radio operators using radios are able to tune to repeaters with VoIP capabilities and use DTMF signals to command the repeater to connect to various other repeaters, thus allowing them to talk to people all around the world, even with "line of sight" VHF radios.
In the U.S., the Federal Communications Commission now requires all interconnected VoIP service providers to comply with requirements comparable to those for traditional telecommunications service providers. VoIP operators in the U.S. are required to support local number portability; make service accessible to people with disabilities; pay regulatory fees, universal service contributions, and other mandated payments; and enable law enforcement authorities to conduct surveillance pursuant to the Communications Assistance for Law Enforcement Act (CALEA). "Interconnected" VoIP operators also must provide Enhanced 911 service, disclose any limitations on their E-911 functionality to their consumers, and obtain affirmative acknowledgements of these disclosures from all consumers. VoIP operators also receive the benefit of certain U.S. telecommunications regulations, including an entitlement to interconnection and exchange of traffic with incumbent local exchange carriers via wholesale carriers. Providers of "nomadic" VoIP service — those who are unable to determine the location of their users — are exempt from state telecommunications regulation.
Throughout the developing world, countries where regulation is weak or captured by the dominant operator, restrictions on the use of VoIP are imposed, including in Panama where VoIP is taxed, Guyana where VoIP is prohibited and India where its retail commercial sales is allowed but only for long distance service. In Ethiopia, where the government is monopolizing telecommunication service, it is a criminal offense to offer services using VoIP. The country has installed firewalls to prevent international calls being made using VoIP. These measures were taken after a popularity in VoIP reduced the income generated by the state owned telecommunication company.
In the European Union, the treatment of VoIP service providers is a decision for each Member State's national telecoms regulator, which must use competition law to define relevant national markets and then determine whether any service provider on those national markets has "significant market power" (and so should be subject to certain obligations). A general distinction is usually made between VoIP services that function over managed networks (via broadband connections) and VoIP services that function over unmanaged networks (essentially, the Internet).
VoIP services that function over managed networks are often considered to be a viable substitute for PSTN telephone services (despite the problems of power outages and lack of geographical information); as a result, major operators that provide these services (in practice, incumbent operators) may find themselves bound by obligations of price control or accounting separation.
VoIP services that function over unmanaged networks are often considered to be too poor in quality to be a viable substitute for PSTN services; as a result, they may be provided without any specific obligations, even if a service provider has "significant market power".
The relevant EU Directive is not clearly drafted concerning obligations which can exist independently of market power (e.g., the obligation to offer access to emergency calls), and it is impossible to say definitively whether VoIP service providers of either type are bound by them. A review of the EU Directive is under way and should be complete by 2007.
In India, it is legal to use VoIP, but it is illegal to have VoIP gateways inside India. This effectively means that people who have PCs can use them to make a VoIP call to any number, but if the remote side is a normal phone, the gateway that converts the VoIP call to a POTS call should not be inside India.
In the Republic of Korea, only providers registered with the government are authorized to offer VoIP services. Unlike many VoIP providers, most of whom offer flat rates, Korean VoIP services are generally metered and charged at rates similar to terrestrial calling. Foreign VoIP providers such as Vonage encounter high barriers to government registration. This issue came to a head in 2006 when internet service providers providing personal internet services by contract to United States Forces Korea members residing on USFK bases threatened to block off access to VoIP services used by USFK members of as an economical way to keep in contact with their families in the United States, on the grounds that the service members' VoIP providers were not registered. A compromise was reached between USFK and Korean telecommunications officials in January 2007, wherein USFK service members arriving in Korea before June 1, 2007 and subscribing to the ISP services provided on base may continue to use their U.S.-based VoIP subscription, but later arrivals must use a Korean-based VoIP provider, which by contract will offer pricing similar to the flat rates offered by U.S. VoIP providers.
IP telephony is basically regulated by Ministry of Internal Affairs and Communications (MIC) as a telecommunication service. The operators have to disclose necessary information on its quality, etc., prior to making contracts with customers, and have an obligation to respond to their complaints cordially.
Many Japanese Internet service providers (ISP) are including IP telephony services. An ISP who also provides IP telephony service is known as a "ITSP (Internet Telephony Service Provider)". Recently, the competition among ITSPs has been activated, by option or set sales, in connection with ADSL or FTTH services.
The tariff system normally applied to Japanese IP telephony is described below;
Between ITSPs, the interconnection is mostly maintained at VoIP level.
High-quality IP telephony is assigned a telephone number, normally starting with the digits 050. When VoIP quality is so high that a customer has difficulty telling the difference between it and a normal telephone, and when the provider relates its number with a location and provides the connection with emergency call capabilities, the provider is allowed to assign a normal telephone number, which is a so-called "0AB-J" number.