Voice over IP
Encyclopedia
Voice over Internet Protocol (Voice over IP, VoIP) is a family of technologies, methodologies, communication protocols, and transmission techniques for the delivery of voice communications and multimedia
sessions over Internet Protocol
(IP) networks, such as the Internet
. Other terms frequently encountered and often used synonymously with VoIP are IP telephony, Internet telephony, voice over broadband (VoBB), broadband telephony, and broadband phone.
Internet telephony refers to communications services—Voice, fax
, SMS
, and/or voice-messaging applications—that are transported via the Internet, rather than the public switched telephone network
(PSTN). The steps involved in originating a VoIP telephone call are signaling and media channel setup, digitization of the analog voice signal, encoding, packetization, and transmission as Internet Protocol
(IP) packets over a packet-switched network. On the receiving side, similar steps (usually in the reverse order) such as reception of the IP packets, decoding of the packets and digital-to-analog conversion reproduce the original voice stream. Even though IP Telephony and VoIP are terms that are used interchangeably, they are actually different; IP telephony has to do with digital telephony systems that use IP protocols for voice communication while VoIP is actually a subset of IP Telephony. VoIP is a technology used by IP telephony as a means of transporting phone calls.
VoIP systems employ session control protocols to control the set-up and tear-down of calls as well as audio codec
s which encode speech allowing transmission over an IP network as digital audio
via an audio stream
. The codec used is varied between different implementations of VoIP (and often a range of codecs are used); some implementations rely on narrowband
and compressed speech
, while others support high fidelity
stereo
codecs.
VoIP is available on many smartphones and internet devices so even the users of portable devices that are not phones can still make calls or send SMS text messages over 3G
or Wi-Fi
.
and open protocols and standards. Examples of technologies used to implement Voice over IP include:
The H.323 protocol was one of the first VoIP protocols that found widespread implementation for long-distance traffic, as well as local area network services. However, since the development of newer, less complex protocols, such as MGCP and SIP, H.323 deployments are increasingly limited to carrying existing long-haul network traffic. In particular, the Session Initiation Protocol (SIP) has gained widespread VoIP market penetration.
A notable proprietary implementation is the Skype protocol
, which is in part based on the principles of peer-to-peer
(P2P) networking.
, by which subscribers place and receive telephone calls in much the same manner as they would via the public switched telephone network
(PSTN). Full-service VoIP phone companies provide inbound and outbound service with Direct Inbound Dialing. Many offer unlimited domestic calling for a flat monthly subscription fee. This sometimes includes international calls to certain countries. Phone calls between subscribers of the same provider are usually free when flat-fee service is not available A VoIP phone is necessary to connect to a VoIP service provider. This can be implemented in several ways:
".
Smartphones and Wi-Fi
enabled mobile phones may have SIP clients built into the firmware or available as an application download.
VoIP solutions aimed at businesses have evolved into "unified communications" services that treat all communications—phone calls, faxes, voice mail, e-mail, Web conferences and more—as discrete units that can all be delivered via any means and to any handset, including cellphones. Two kinds of competitors are competing in this space: one set is focused on VoIP for medium to large enterprises, while another is targeting the small-to-medium business (SMB) market.
VoIP allows both voice and data communications to be run over a single network, which can significantly reduce infrastructure costs.
The prices of extensions on VoIP are lower than for PBX and key systems. VoIP switches may run on commodity hardware, such as PCs or Linux systems. Rather than closed architectures, these devices rely on standard interfaces.
VoIP devices have simple, intuitive user interfaces, so users can often make simple system configuration changes. Dual-mode phones enable users to continue their conversations as they move between an outside cellular service and an internal Wi-Fi
network, so that it is no longer necessary to carry both a desktop phone and a cellphone. Maintenance becomes simpler as there are fewer devices to oversee.
Skype
, which originally marketed itself as a service among friends, has begun to cater to businesses, providing free-of-charge connections between any users on the Skype network and connecting to and from ordinary PSTN telephones for a charge.
In the United States the Social Security Administration (SSA) is converting its field offices of 63,000 workers from traditional phone installations to a VoIP infrastructure carried over its existing data network.
Examples include:
(QoS) guarantees. Therefore, VoIP implementations may face problems mitigating latency
and jitter
.
By default, network routers handle traffic on a first-come, first-served basis. Network routers on high volume traffic links may introduce latency that exceeds permissible thresholds for VoIP. Fixed delays cannot be controlled, as they are caused by the physical distance the packets travel; however, latency can be minimized by marking voice packets as being delay-sensitive with methods such as DiffServ.
A VoIP packet usually has to wait for the current packet to finish transmission, although it is possible to preempt (abort) a less important packet in mid-transmission, although this is not commonly done, especially on high-speed links where transmission times are short even for maximum-sized packets. An alternative to preemption on slower links, such as dialup and DSL, is to reduce the maximum transmission time by reducing the maximum transmission unit
. But every packet must contain protocol headers, so this increases relative header overhead on every link along the user's Internet paths, not just the bottleneck (usually Internet access) link.
ADSL modems provide Ethernet (or Ethernet over USB) connections to local equipment, but inside they are actually Asynchronous Transfer Mode
(ATM) modems. They use ATM Adaptation Layer 5
(AAL5) to segment each Ethernet packet into a series of 53-byte ATM cells for transmission and reassemble them back into Ethernet packets at the receiver. A virtual circuit identifier
(VCI) is part of the 5-byte header on every ATM cell, so the transmitter can multiplex
the active virtual circuits (VCs) in any arbitrary order. Cells from the same VC are always sent sequentially.
However, the great majority of DSL providers use only one VC for each customer, even those with bundled VoIP service. Every Ethernet packet must be completely transmitted before another can begin. If a second PVC were established, given high priority and reserved for VoIP, then a low priority data packet could be suspended in mid-transmission and a VoIP packet sent right away on the high priority VC. Then the link would pick up the low priority VC where it left off. Because ATM links are multiplexed on a cell-by-cell basis, a high priority packet would have to wait at most 53 byte times to begin transmission. There would be no need to reduce the interface MTU and accept the resulting increase in higher layer protocol overhead, and no need to abort a low priority packet and resend it later.
ATM has substantial header overhead: 5/53 = 9.4%, roughly twice the total header overhead of a 1500 byte TCP/IP Ethernet packet (with TCP timestamps). This "ATM tax" is incurred by every DSL user whether or not he takes advantage of multiple virtual circuits - and few can.
ATM's potential for latency reduction is greatest on slow links, because worst-case latency decreases with increasing link speed. A full-size (1500 byte) Ethernet frame takes 94 ms to transmit at 128 kbit/s but only 8 ms at 1.5 Mbit/s. If this is the bottleneck link, this latency is probably small enough to ensure good VoIP performance without MTU reductions or multiple ATM PVCs. The latest generations of DSL, VDSL and VDSL2, carry Ethernet without intermediate ATM/AAL5 layers, and they generally support IEEE 802.1p priority tagging so that VoIP can be queued ahead of less time-critical traffic.
Voice, and all other data, travels in packets over IP networks with fixed maximum capacity. This system may be more prone to congestion and DoS attack
s than traditional circuit switched systems; a circuit switched system of insufficient capacity will refuse new connections while carrying the remainder without impairment, while the quality of real-time data such as telephone conversations on packet-switched networks degrades dramatically.
Fixed delays cannot be controlled as they are caused by the physical distance the packets travel. They are especially problematic when satellite circuits are involved because of the long distance to a geostationary satellite
and back; delays of 400–600 ms are typical.
When the load on a link grows so quickly that its switches experience queue overflows, congestion results and data packets are lost. This signals a transport protocol like TCP
to reduce its transmission rate to alleviate the congestion. But VoIP usually uses UDP
not TCP because recovering from congestion through retransmission usually entails too much latency. So QoS mechanisms can avoid the undesirable loss of VoIP packets by immediately transmitting them ahead of any queued bulk traffic on the same link, even when that bulk traffic queue is overflowing.
The receiver must resequence IP packets that arrive out of order and recover gracefully when packets arrive too late or not at all. Jitter
results from the rapid and random (i.e., unpredictable) changes in queue lengths along a given Internet path due to competition from other users for the same transmission links. VoIP receivers counter jitter by storing incoming packets briefly in a "de-jitter" or "playout" buffer, deliberately increasing latency to improve the chance that each packet will be on hand when it is time for the voice engine
to play it. The added delay is thus a compromise between excessive latency and excessive dropout
, i.e., momentary audio interruptions.
Although jitter is a random variable, it is the sum of several other random variables that are at least somewhat independent: the individual queuing delays of the routers along the Internet path in question. Thus according to the central limit theorem
, we can model jitter as a gaussian random variable. This suggests continually estimating the mean delay and its standard deviation and setting the playout delay so that only packets delayed more than several standard deviations above the mean will arrive too late to be useful. In practice, however, the variance in latency of many Internet paths is dominated by a small number (often one) of relatively slow and congested "bottleneck" links. Most Internet backbone links are now so fast (e.g. 10 Gbit/s) that their delays are dominated by the transmission medium (e.g. optical fiber) and the routers driving them do not have enough buffering for queuing delays to be significant.
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 code
s 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 Extended Report (RFC 3611), SIP
RTCP Summary Reports, H.460.9 Annex B (for H.323
), H.248.30
and MGCP
extensions. The RFC 3611 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 (because of jitter), packet loss/discard burst metrics (burst length/density, gap length/density), network delay, end system delay, signal / noise / echo level, Mean Opinion Score
s (MOS) and R factors and configuration information related to the jitter buffer.
RFC 3611 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. RFC 3611 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.
and physical layer
include quality-of-service mechanisms that can be used to ensure that applications like VoIP work well even in congested scenarios. Some examples include:
IP Phone
s and VoIP telephone adapters connect to routers or cable modem
s which typically depend on the availability of mains electricity
or locally generated power. Some VoIP service providers use customer premise equipment (e.g., cablemodems) with battery-backed power supplies to assure uninterrupted service for up to several hours in case of local power failures. Such battery-backed devices typically are designed for use with analog handsets.
Some VoIP service providers implement services to route calls to other telephone services of the subscriber, such a cellular phone, in the event that the customer's network device is inaccessible to terminate the call.
The susceptibility of phone service to power failures is a common problem even with traditional analog service in areas where many customers purchase modern telephone units that operate with wireless handsets to a base station, or that have other modern phone features, such as built-in voicemail or phone book features.
makes it difficult to locate network users geographically. Emergency call
s, therefore, cannot easily be routed to a nearby call center. Sometimes, VoIP systems may route emergency calls to a non-emergency phone line at the intended department; in the United States, at least one major police department has strongly objected to this practice as potentially endangering the public.
A fixed line phone has a direct relationship between a telephone number and a physical location. If an emergency call comes from that number, then the physical location is known.
In the IP world, it is not so simple. A broadband provider may know the location where the wires terminate, but this does not necessarily allow the mapping of an IP address to that location. IP addresses are often dynamically assigned, so the ISP may allocate an address for online access, or at the time a broadband router is engaged. The ISP recognizes individual IP addresses, but does not necessarily know to which physical location it corresponds. The broadband service provider knows the physical location, but is not necessarily tracking the IP addresses in use.
There are more complications since IP allows a great deal of mobility. For example, a broadband connection can be used to dial a virtual private network that is employer-owned. When this is done, the IP address being used will belong to the range of the employer, rather than the address of the ISP, so this could be many kilometres away or even in another country. To provide another example: if mobile data is used, e.g., a 3G mobile handset or USB wireless broadband adapter, then the IP address has no relationship with any physical location, since a mobile user could be anywhere that there is network coverage, even roaming via another cellular company.
In short, there is no relationship between IP address and physical location, so the address itself reveals no useful information for the emergency services.
At the VoIP level, a phone or gateway may identify itself with a SIP
registrar by using a username and password. So in this case, the Internet Telephony Service Provider (ITSP) knows that a particular user is online, and can relate a specific telephone number to the user. However, it does not recognize how that IP traffic was engaged. Since the IP address itself does not necessarily provide location information presently, today a "best efforts" approach is to use an available database to find that user and the physical address the user chose to associate with that telephone number—clearly an imperfect solution.
VoIP Enhanced 911
(E911) is a method by which VoIP providers in the United States 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 VoIP providers that provide access to the public switched telephone network are required to implement E911, a service for which the subscriber may be charged. Participation in E911 is not required and customers may opt-out of E911 service.
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.
and equivalent services in other locales. However, the internet as designed by DARPA in the early 1980s was specifically designed to be fault tolerant under adverse conditions. Even during the 9/11 attacks on the World Trade Centers the internet routed data around the failed nodes that were housed in or near the towers. So single point failures while possible in some geographic areas are not the norm for the internet as a whole.
(LNP) and Mobile number portability
(MNP) also impact VoIP business. 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. Number portability is a service that allows a subscriber to select a new telephone carrier without requiring a new number to be issued. Typically, it is the responsibility of the former carrier to "map" the old number to the undisclosed number assigned by the new carrier. This is achieved by maintaining a database of numbers. A dialed number is initially received by the original carrier and quickly rerouted to the new carrier. Multiple porting references must be maintained even if the subscriber returns to the original carrier. The FCC mandates carrier compliance with these consumer-protection stipulations.
A voice call originating in the VoIP environment also faces challenges to reach its destination if the number is routed to a mobile phone number on a traditional mobile carrier. VoIP has been identified in the past 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. This rating is subject to some debate given the complexity of call routing created by number portability. 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 must now determine 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 which home network a mobile phone number belongs to. As the popularity of VoIP increases 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 by assuring that the voice call will actually work, VoIP service providers are able to offer business subscribers the level of reliability they require.
is a global FGFnumbering standard for both the PSTN
and PLMN
. Most VoIP implementations support E.164
to allow calls to be routed to and from VoIP subscribers and the PSTN/PLMN. VoIP implementations can also allow other identification techniques to be used. For example, Skype allows subscribers to choose "Skype names" (usernames) whereas SIP implementations can use URI
s similar to email addresses
. Often VoIP implementations employ methods of translating non-E.164 identifiers to E.164 numbers and vice-versa, such as the Skype-In service provided by Skype and the ENUM
service in IMS and SIP.
Echo can also be an issue for PSTN integration. Common causes of echo include impedance mismatches
in analog circuitry and acoustic coupling of the transmit and receive signal at the receiving end.
s who know about these vulnerabilities (such as insecure passwords) can institute denial-of-service attacks, harvest customer data, record conversations and break into voice mailboxes.
Another challenge is routing VoIP traffic through firewalls and network address translators. Private Session Border Controller
s are used along with firewalls to enable VoIP calls to and from protected networks. For example, 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 NATs involve using protocols such as STUN
or Interactive Connectivity Establishment
(ICE).
Many consumer VoIP solutions do not support encryption, although having a secure phone is much easier to implement with VoIP than traditional phone lines. As a result, it is relatively easy to eavesdrop on VoIP calls and even change their content. An attacker with a packet sniffer could intercept your VoIP calls if you are not on a secure VLAN. However, physical security of the switches within an enterprise and the facility security provided by ISPs make packet capture less of a problem than originally foreseen. Further research has shown that tapping into a fiber optic network without detection is difficult if not impossible. This means that once a voice packet is within the internet backbone it is relatively safe from interception.
There are open source solutions, such as Wireshark
, that facilitate sniffing of VoIP conversations. A modicum of security is afforded by patented audio codecs in proprietary implementations that are not easily available for open source applications; however, such security through obscurity
has not proven effective in other fields. Some vendors also use compression, which may make eavesdropping
more difficult. However, real security requires encryption and cryptographic authentication which are not widely supported at a consumer level. The existing security standard Secure Real-time Transport Protocol
(SRTP) and the new ZRTP
protocol are available on Analog Telephone Adapters (ATAs)
as well as various softphone
s. It is possible to use IPsec
to secure P2P VoIP by using opportunistic encryption
. Skype does not use SRTP, but uses encryption which is transparent to the Skype provider. In 2005, Skype invited a researcher, Dr Tom Berson, to assess the security of the Skype software, and his conclusions are available in a published report.
. Secure Voice over Secure IP is accomplished by using Type 1 encryption
on a classified network, like SIPRNet
. Public Secure VoIP is also available with free GNU programs.
support among VoIP providers varies, but is provided by the majority of VoIP providers.
Many VoIP carriers allow callers to configure arbitrary Caller ID information, thus permitting spoofing attack
s. Business grade VoIP equipment and software often makes it easy to modify caller ID information, providing many businesses great flexibility.
The Truth in Caller ID Act has been in preparation in the US Congress since 2006, but as of January 2009 still has not been enacted. This bill proposes to make it a crime in the United States to "knowingly transmit misleading or inaccurate caller identification information with the intent to defraud, cause harm, or wrongfully obtain anything of value ..."
s using the touch-tone system. The VoIP user may use a pulse-to-tone converter, if needed.
s are not designed for fax transmission; they are designed to digitize an analog representation of a human voice efficiently. However, the inefficiency of digitizing an analog representation (modem signal) of a digital representation (a document image) of analog data (an original document) more than negates any bandwidth advantage of VoIP. In other words, the fax "sounds" simply do not fit in the VoIP channel. An alternative IP-based solution for delivering fax-over-IP called T.38
is available. Sending faxes using VoIP is sometimes referred to as FoIP, or Fax over IP.
The T.38 protocol is designed to compensate for the differences between traditional packet-less communications over analog lines and packet based transmissions which are the basis for IP communications. 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. TCP is better suited for use between two IP devices. However, older fax machines, connected to an analog system, benefit from UDP near real-time characteristics due to the "no recovery rule" when a UDP packet is lost or an error occurs during transmission. UDP transmissions are preferred as they do not require testing for dropped packets and as such since each T.38 packet transmission includes a majority of the data sent in the prior packet, a T.38 termination point has a higher degree of success in re-assembling the fax transmission back into its original form for interpretation by the end device. This in an attempt to overcome the obstacles of simulating real time transmissions using packet based protocol.
There have been updated versions of T.30 to resolve the fax over IP issues, which is the core fax protocol. Some newer high end fax machines have T.38 built-in capabilities which allow the user to plug right into the network and transmit/receive faxes in native T.38 like the Ricoh 4410NF Fax Machine. A unique feature of T.38 is that each packet contains a portion of the main data sent in the previous packet. With T.38, two successive lost packets are needed 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 an increased likelihood that you will receive enough of the transmission to satisfy the requirements of the fax machine for output of the sent document.
s, satellite television
receivers, alarm
systems, conventional modem
s and other similar devices that depend on access to a PSTN telephone
line for some or all of their functionality.
These types of calls sometimes complete without any problems, but in other cases they fail. 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 PSTN telephone line would be available in consumer's homes.
Another legal issue that the US Congress is debating concerns changes to the Foreign Intelligence Surveillance Act. The issue in question is calls between Americans and foreigners. The National Security Agency (NSA) is not authorized to tap Americans' conversations without a warrant—but the Internet, and specifically VoIP does not draw as clear a line to the location of a caller or a call's recipient as the traditional phone system does. As VoIP's low cost and flexibility convinces more and more organizations to adopt the technology, the surveillance for law enforcement agencies becomes more difficult. VoIP technology has also increased security concerns because VoIP and similar technologies have made it more difficult for the government to determine where a target is physically located when communications are being intercepted, and that creates a whole set of new legal challenges.
In the US, 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 US 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 US telecommunications regulations, including an entitlement to interconnection
and exchange of traffic with incumbent local exchange carrier
s 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 the popularity of 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 UAE
and Oman it is illegal to use any form of VoIP, to the extent that Web sites of Skype
and Gizmo5
are blocked. Providing or using VoIP services is illegal in Oman. Those who violate the law stand to be fined 50,000 Omani Rial (about 130,317 US dollars) or spend two years in jail or both. In 2009, police in Oman have raided 121 internet cafes throughout the country and arrested 212 people for using/providing VoIP services.
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 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 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 US-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 US VoIP providers.
Multimedia
Multimedia is media and content that uses a combination of different content forms. The term can be used as a noun or as an adjective describing a medium as having multiple content forms. The term is used in contrast to media which use only rudimentary computer display such as text-only, or...
sessions over Internet Protocol
Internet Protocol
The Internet Protocol is the principal communications protocol used for relaying datagrams across an internetwork using the Internet Protocol Suite...
(IP) networks, such as the Internet
Internet
The Internet is a global system of interconnected computer networks that use the standard Internet protocol suite to serve billions of users worldwide...
. Other terms frequently encountered and often used synonymously with VoIP are IP telephony, Internet telephony, voice over broadband (VoBB), broadband telephony, and broadband phone.
Internet telephony refers to communications services—Voice, fax
Fax
Fax , sometimes called telecopying, is the telephonic transmission of scanned printed material , normally to a telephone number connected to a printer or other output device...
, SMS
SMS
SMS is a form of text messaging communication on phones and mobile phones. The terms SMS or sms may also refer to:- Computer hardware :...
, and/or voice-messaging applications—that are transported via the Internet, rather than the public switched telephone network
Public switched telephone network
The public switched telephone network is the network of the world's public circuit-switched telephone networks. It consists of telephone lines, fiber optic cables, microwave transmission links, cellular networks, communications satellites, and undersea telephone cables, all inter-connected by...
(PSTN). The steps involved in originating a VoIP telephone call are signaling and media channel setup, digitization of the analog voice signal, encoding, packetization, and transmission as Internet Protocol
Internet Protocol
The Internet Protocol is the principal communications protocol used for relaying datagrams across an internetwork using the Internet Protocol Suite...
(IP) packets over a packet-switched network. On the receiving side, similar steps (usually in the reverse order) such as reception of the IP packets, decoding of the packets and digital-to-analog conversion reproduce the original voice stream. Even though IP Telephony and VoIP are terms that are used interchangeably, they are actually different; IP telephony has to do with digital telephony systems that use IP protocols for voice communication while VoIP is actually a subset of IP Telephony. VoIP is a technology used by IP telephony as a means of transporting phone calls.
VoIP systems employ session control protocols to control the set-up and tear-down of calls as well as audio codec
Audio codec
All codecs are devices or computer programs capable of coding or decoding a digital data stream or signal.The term audio codec has two meanings depending on the context:...
s which encode speech allowing transmission over an IP network as digital audio
Digital audio
Digital audio is sound reproduction using pulse-code modulation and digital signals. Digital audio systems include analog-to-digital conversion , digital-to-analog conversion , digital storage, processing and transmission components...
via an audio stream
Streaming media
Streaming media is multimedia that is constantly received by and presented to an end-user while being delivered by a streaming provider.The term "presented" is used in this article in a general sense that includes audio or video playback. The name refers to the delivery method of the medium rather...
. The codec used is varied between different implementations of VoIP (and often a range of codecs are used); some implementations rely on narrowband
Narrowband
In radio, narrowband describes a channel in which the bandwidth of the message does not significantly exceed the channel's coherence bandwidth. It is a common misconception that narrowband refers to a channel which occupies only a "small" amount of space on the radio spectrum.The opposite of...
and compressed speech
Speech coding
Speech coding is the application of data compression of digital audio signals containing speech. Speech coding uses speech-specific parameter estimation using audio signal processing techniques to model the speech signal, combined with generic data compression algorithms to represent the resulting...
, while others support high fidelity
High fidelity
High fidelity—or hi-fi—reproduction is a term used by home stereo listeners and home audio enthusiasts to refer to high-quality reproduction of sound or images, to distinguish it from the poorer quality sound produced by inexpensive audio equipment...
stereo
Stereophonic sound
The term Stereophonic, commonly called stereo, sound refers to any method of sound reproduction in which an attempt is made to create an illusion of directionality and audible perspective...
codecs.
VoIP is available on many smartphones and internet devices so even the users of portable devices that are not phones can still make calls or send SMS text messages over 3G
3G
3G or 3rd generation mobile telecommunications is a generation of standards for mobile phones and mobile telecommunication services fulfilling the International Mobile Telecommunications-2000 specifications by the International Telecommunication Union...
or Wi-Fi
Wi-Fi
Wi-Fi or Wifi, is a mechanism for wirelessly connecting electronic devices. A device enabled with Wi-Fi, such as a personal computer, video game console, smartphone, or digital audio player, can connect to the Internet via a wireless network access point. An access point has a range of about 20...
.
Protocols
Voice over IP has been implemented in various ways using both proprietaryProprietary protocol
In telecommunications, a proprietary protocol is a communications protocol owned by a single organization or individual.-Enforcement:Proprietors may enforce restrictions through patents and by keeping the protocol specification a trade secret...
and open protocols and standards. Examples of technologies used to implement Voice over IP include:
- H.323H.323H.323 is a recommendation from the ITU Telecommunication Standardization Sector that defines the protocols to provide audio-visual communication sessions on any packet network...
- IP Multimedia SubsystemIP Multimedia SubsystemThe IP Multimedia Subsystem or IP Multimedia Core Network Subsystem is an architectural framework for delivering Internet Protocol multimedia services. It was originally designed by the wireless standards body 3rd Generation Partnership Project , as a part of the vision for evolving mobile...
(IMS) - Media Gateway Control Protocol (MGCP)Media Gateway Control Protocol (MGCP)MGCP is an implementation of the Media Gateway Control Protocol architecture for controlling media gateways on Internet Protocol networks and the public switched telephone network . The general base architecture and programming interface is described in RFC 2805 and the current specific MGCP...
- Session Initiation ProtocolSession Initiation ProtocolThe Session Initiation Protocol is an IETF-defined signaling protocol widely used for controlling communication sessions such as voice and video calls over Internet Protocol . The protocol can be used for creating, modifying and terminating two-party or multiparty sessions...
(SIP) - Real-time Transport ProtocolReal-time Transport ProtocolThe Real-time Transport Protocol defines a standardized packet format for delivering audio and video over IP networks. RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications, television services and...
(RTP) - Session Description ProtocolSession Description ProtocolThe Session Description Protocol is a format for describing streaming media initialization parameters. The IETF published the original specification as an IETF Proposed Standard in April 1998, and subsequently published a revised specification as an IETF Proposed Standard as RFC 4566 in July...
(SDP) - Inter-Asterisk eXchange (IAX)
The H.323 protocol was one of the first VoIP protocols that found widespread implementation for long-distance traffic, as well as local area network services. However, since the development of newer, less complex protocols, such as MGCP and SIP, H.323 deployments are increasingly limited to carrying existing long-haul network traffic. In particular, the Session Initiation Protocol (SIP) has gained widespread VoIP market penetration.
A notable proprietary implementation is the Skype protocol
Skype Protocol
The Skype protocol is a proprietary Internet telephony network based on peer-to-peer architecture, used by Skype. The protocol's specifications have not been made publicly available by Skype and official applications using the protocol are closed-source....
, which is in part based on the principles of peer-to-peer
Peer-to-peer
Peer-to-peer computing or networking is a distributed application architecture that partitions tasks or workloads among peers. Peers are equally privileged, equipotent participants in the application...
(P2P) networking.
Consumer market
A major development that started in 2004 was the introduction of mass-market VoIP services that utilize existing broadband Internet accessBroadband Internet access
Broadband Internet access, often shortened to just "broadband", is a high data rate, low-latency connection to the Internet— typically contrasted with dial-up access using a 56 kbit/s modem or satellite Internet with inherently high latency....
, by which subscribers place and receive telephone calls in much the same manner as they would via the public switched telephone network
Public switched telephone network
The public switched telephone network is the network of the world's public circuit-switched telephone networks. It consists of telephone lines, fiber optic cables, microwave transmission links, cellular networks, communications satellites, and undersea telephone cables, all inter-connected by...
(PSTN). Full-service VoIP phone companies provide inbound and outbound service with Direct Inbound Dialing. Many offer unlimited domestic calling for a flat monthly subscription fee. This sometimes includes international calls to certain countries. Phone calls between subscribers of the same provider are usually free when flat-fee service is not available A VoIP phone is necessary to connect to a VoIP service provider. This can be implemented in several ways:
- Dedicated VoIP phones connect directly to the IP network using technologies such as wired EthernetEthernetEthernet is a family of computer networking technologies for local area networks commercially introduced in 1980. Standardized in IEEE 802.3, Ethernet has largely replaced competing wired LAN technologies....
or wireless Wi-FiWi-FiWi-Fi or Wifi, is a mechanism for wirelessly connecting electronic devices. A device enabled with Wi-Fi, such as a personal computer, video game console, smartphone, or digital audio player, can connect to the Internet via a wireless network access point. An access point has a range of about 20...
. They are typically designed in the style of traditional digital business telephones. - An analog telephone adapter is a device that connects to the network and implements the electronics and firmware to operate a conventional analog telephone attached through a modular phone jack. Some residential Internet gateways and cablemodems have this function built in.
- A softphoneSoftphoneA softphone is a software program for making telephone calls over the Internet using a general purpose computer, rather than using dedicated hardware. Often a softphone is designed to behave like a traditional telephone, sometimes appearing as an image of a phone, with a display panel and buttons...
is application software installed on a networked computerComputerA computer is a programmable machine designed to sequentially and automatically carry out a sequence of arithmetic or logical operations. The particular sequence of operations can be changed readily, allowing the computer to solve more than one kind of problem...
that is equipped with a microphone and speaker, or headset. The application typically presents a dial pad and display field to the user to operate the application by mouse clicks or keyboard input.
PSTN and mobile network providers
It is becoming increasingly common for telecommunications providers to use VoIP telephony over dedicated and public IP networks to connect switching centres and to interconnect with other telephony network providers; this is often referred to as "IP backhaulBackhaul (telecommunications)
In a hierarchical telecommunications network the backhaul portion of the network comprises the intermediate links between the core network, or backbone, of the network and the small subnetworks at the "edge" of the entire hierarchical network...
".
Smartphones and Wi-Fi
Wi-Fi
Wi-Fi or Wifi, is a mechanism for wirelessly connecting electronic devices. A device enabled with Wi-Fi, such as a personal computer, video game console, smartphone, or digital audio player, can connect to the Internet via a wireless network access point. An access point has a range of about 20...
enabled mobile phones may have SIP clients built into the firmware or available as an application download.
Corporate use
Because of the bandwidth efficiency and low costs that VoIP technology can provide, businesses are migrating from traditional copper-wire telephone systems to VoIP systems to reduce their monthly phone costs. In 2008, 80% of all new PBX lines installed internationally were VoIP.VoIP solutions aimed at businesses have evolved into "unified communications" services that treat all communications—phone calls, faxes, voice mail, e-mail, Web conferences and more—as discrete units that can all be delivered via any means and to any handset, including cellphones. Two kinds of competitors are competing in this space: one set is focused on VoIP for medium to large enterprises, while another is targeting the small-to-medium business (SMB) market.
VoIP allows both voice and data communications to be run over a single network, which can significantly reduce infrastructure costs.
The prices of extensions on VoIP are lower than for PBX and key systems. VoIP switches may run on commodity hardware, such as PCs or Linux systems. Rather than closed architectures, these devices rely on standard interfaces.
VoIP devices have simple, intuitive user interfaces, so users can often make simple system configuration changes. Dual-mode phones enable users to continue their conversations as they move between an outside cellular service and an internal Wi-Fi
Wi-Fi
Wi-Fi or Wifi, is a mechanism for wirelessly connecting electronic devices. A device enabled with Wi-Fi, such as a personal computer, video game console, smartphone, or digital audio player, can connect to the Internet via a wireless network access point. An access point has a range of about 20...
network, so that it is no longer necessary to carry both a desktop phone and a cellphone. Maintenance becomes simpler as there are fewer devices to oversee.
Skype
Skype
Skype is a software application that allows users to make voice and video calls and chat over the Internet. Calls to other users within the Skype service are free, while calls to both traditional landline telephones and mobile phones can be made for a fee using a debit-based user account system...
, which originally marketed itself as a service among friends, has begun to cater to businesses, providing free-of-charge connections between any users on the Skype network and connecting to and from ordinary PSTN telephones for a charge.
In the United States the Social Security Administration (SSA) is converting its field offices of 63,000 workers from traditional phone installations to a VoIP infrastructure carried over its existing data network.
Operational cost
VoIP can be a benefit for reducing communication and infrastructure costs.Examples include:
- Routing phone calls over existing data networks to avoid the need for separate voice and data networks.
- The ability to transmit more than one telephone call over a single broadband connection.
- Secure calls using standardized protocols (such as Secure Real-time Transport ProtocolSecure Real-time Transport ProtocolThe Secure Real-time Transport Protocol defines a profile of RTP , intended to provide encryption, message authentication and integrity, and replay protection to the RTP data in both unicast and multicast applications...
). Most of the difficulties of creating a secure telephoneSecure telephoneA secure telephone is a telephone that provides voice security in the form of end-to-end encryption for the telephone call, and in some cases also the mutual authentication of the call parties, protecting them against a man-in-the-middle attack...
connection over traditional phone lines, such as digitizing and digital transmission, are already in place with VoIP. It is only necessary to encrypt and authenticate the existing data stream.
Quality of service
Communication on the IP network is inherently less reliable in contrast to the circuit-switched public telephone network, as it does not provide a network-based mechanism to ensure that data packets are not lost, and are delivered in sequential order. It is a best-effort network without fundamental Quality of ServiceQuality of service
The quality of service refers to several related aspects of telephony and computer networks that allow the transport of traffic with special requirements...
(QoS) guarantees. Therefore, VoIP implementations may face problems mitigating latency
Latency (engineering)
Latency is a measure of time delay experienced in a system, the precise definition of which depends on the system and the time being measured. Latencies may have different meaning in different contexts.-Packet-switched networks:...
and jitter
Jitter
Jitter is the undesired deviation from true periodicity of an assumed periodic signal in electronics and telecommunications, often in relation to a reference clock source. Jitter may be observed in characteristics such as the frequency of successive pulses, the signal amplitude, or phase of...
.
By default, network routers handle traffic on a first-come, first-served basis. Network routers on high volume traffic links may introduce latency that exceeds permissible thresholds for VoIP. Fixed delays cannot be controlled, as they are caused by the physical distance the packets travel; however, latency can be minimized by marking voice packets as being delay-sensitive with methods such as DiffServ.
A VoIP packet usually has to wait for the current packet to finish transmission, although it is possible to preempt (abort) a less important packet in mid-transmission, although this is not commonly done, especially on high-speed links where transmission times are short even for maximum-sized packets. An alternative to preemption on slower links, such as dialup and DSL, is to reduce the maximum transmission time by reducing the maximum transmission unit
Maximum transmission unit
In computer networking, the maximum transmission unit of a communications protocol of a layer is the size of the largest protocol data unit that the layer can pass onwards. MTU parameters usually appear in association with a communications interface...
. But every packet must contain protocol headers, so this increases relative header overhead on every link along the user's Internet paths, not just the bottleneck (usually Internet access) link.
ADSL modems provide Ethernet (or Ethernet over USB) connections to local equipment, but inside they are actually Asynchronous Transfer Mode
Asynchronous Transfer Mode
Asynchronous Transfer Mode is a standard switching technique designed to unify telecommunication and computer networks. It uses asynchronous time-division multiplexing, and it encodes data into small, fixed-sized cells. This differs from approaches such as the Internet Protocol or Ethernet that...
(ATM) modems. They use ATM Adaptation Layer 5
ATM Adaptation Layer 5
ATM Adaptation Layer 5 is an ATM adaptation layer used to send variable-length packets up to 65,535 octets in size across an Asynchronous Transfer Mode network....
(AAL5) to segment each Ethernet packet into a series of 53-byte ATM cells for transmission and reassemble them back into Ethernet packets at the receiver. A virtual circuit identifier
Virtual Circuit Identifier
A Virtual Channel Identifier is a unique identifier which indicates a particular virtual circuit on a network. It is a 16-bit field in the header of an ATM cell...
(VCI) is part of the 5-byte header on every ATM cell, so the transmitter can multiplex
Multiplexing
The multiplexed signal is transmitted over a communication channel, which may be a physical transmission medium. The multiplexing divides the capacity of the low-level communication channel into several higher-level logical channels, one for each message signal or data stream to be transferred...
the active virtual circuits (VCs) in any arbitrary order. Cells from the same VC are always sent sequentially.
However, the great majority of DSL providers use only one VC for each customer, even those with bundled VoIP service. Every Ethernet packet must be completely transmitted before another can begin. If a second PVC were established, given high priority and reserved for VoIP, then a low priority data packet could be suspended in mid-transmission and a VoIP packet sent right away on the high priority VC. Then the link would pick up the low priority VC where it left off. Because ATM links are multiplexed on a cell-by-cell basis, a high priority packet would have to wait at most 53 byte times to begin transmission. There would be no need to reduce the interface MTU and accept the resulting increase in higher layer protocol overhead, and no need to abort a low priority packet and resend it later.
ATM has substantial header overhead: 5/53 = 9.4%, roughly twice the total header overhead of a 1500 byte TCP/IP Ethernet packet (with TCP timestamps). This "ATM tax" is incurred by every DSL user whether or not he takes advantage of multiple virtual circuits - and few can.
ATM's potential for latency reduction is greatest on slow links, because worst-case latency decreases with increasing link speed. A full-size (1500 byte) Ethernet frame takes 94 ms to transmit at 128 kbit/s but only 8 ms at 1.5 Mbit/s. If this is the bottleneck link, this latency is probably small enough to ensure good VoIP performance without MTU reductions or multiple ATM PVCs. The latest generations of DSL, VDSL and VDSL2, carry Ethernet without intermediate ATM/AAL5 layers, and they generally support IEEE 802.1p priority tagging so that VoIP can be queued ahead of less time-critical traffic.
Voice, and all other data, travels in packets over IP networks with fixed maximum capacity. This system may be more prone to congestion and DoS attack
Denial-of-service attack
A denial-of-service attack or distributed denial-of-service attack is an attempt to make a computer resource unavailable to its intended users...
s than traditional circuit switched systems; a circuit switched system of insufficient capacity will refuse new connections while carrying the remainder without impairment, while the quality of real-time data such as telephone conversations on packet-switched networks degrades dramatically.
Fixed delays cannot be controlled as they are caused by the physical distance the packets travel. They are especially problematic when satellite circuits are involved because of the long distance to a geostationary satellite
Geosynchronous satellite
A geosynchronous Satellite is a satellite whose orbit on the Earth repeats regularly over points on the Earth over time. If such a satellite's orbit lies over the equator, the orbit is circular and its angular velocity is the same as the earth's, then it is called a geostationary satellite...
and back; delays of 400–600 ms are typical.
When the load on a link grows so quickly that its switches experience queue overflows, congestion results and data packets are lost. This signals a transport protocol like TCP
Transmission Control Protocol
The Transmission Control Protocol is one of the core protocols of the Internet Protocol Suite. TCP is one of the two original components of the suite, complementing the Internet Protocol , and therefore the entire suite is commonly referred to as TCP/IP...
to reduce its transmission rate to alleviate the congestion. But VoIP usually uses UDP
User Datagram Protocol
The User Datagram Protocol is one of the core members of the Internet Protocol Suite, the set of network protocols used for the Internet. With UDP, computer applications can send messages, in this case referred to as datagrams, to other hosts on an Internet Protocol network without requiring...
not TCP because recovering from congestion through retransmission usually entails too much latency. So QoS mechanisms can avoid the undesirable loss of VoIP packets by immediately transmitting them ahead of any queued bulk traffic on the same link, even when that bulk traffic queue is overflowing.
The receiver must resequence IP packets that arrive out of order and recover gracefully when packets arrive too late or not at all. Jitter
Jitter
Jitter is the undesired deviation from true periodicity of an assumed periodic signal in electronics and telecommunications, often in relation to a reference clock source. Jitter may be observed in characteristics such as the frequency of successive pulses, the signal amplitude, or phase of...
results from the rapid and random (i.e., unpredictable) changes in queue lengths along a given Internet path due to competition from other users for the same transmission links. VoIP receivers counter jitter by storing incoming packets briefly in a "de-jitter" or "playout" buffer, deliberately increasing latency to improve the chance that each packet will be on hand when it is time for the voice engine
Voice engine
A voice engine is a software subsystem for bidirectional audio communication, typically used as part of a telecommunications system to simulate a telephone. It functions like a data pump for audio data, specifically voice data...
to play it. The added delay is thus a compromise between excessive latency and excessive dropout
Dropout (electronics)
Dropout within the realm of electronics and electrical engineering, has a number of uses.It is the dropping away of a flake of magnetic material from magnetic tape, leading to loss of signal, or a failure to properly read a binary character from data storage...
, i.e., momentary audio interruptions.
Although jitter is a random variable, it is the sum of several other random variables that are at least somewhat independent: the individual queuing delays of the routers along the Internet path in question. Thus according to the central limit theorem
Central limit theorem
In probability theory, the central limit theorem states conditions under which the mean of a sufficiently large number of independent random variables, each with finite mean and variance, will be approximately normally distributed. The central limit theorem has a number of variants. In its common...
, we can model jitter as a gaussian random variable. This suggests continually estimating the mean delay and its standard deviation and setting the playout delay so that only packets delayed more than several standard deviations above the mean will arrive too late to be useful. In practice, however, the variance in latency of many Internet paths is dominated by a small number (often one) of relatively slow and congested "bottleneck" links. Most Internet backbone links are now so fast (e.g. 10 Gbit/s) that their delays are dominated by the transmission medium (e.g. optical fiber) and the routers driving them do not have enough buffering for queuing delays to be significant.
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
Capillary routing
In networking and in graph theory, capillary routing, for a given network, is a multi-path solution between a pair of source and destination nodes...
it has been suggested to use at the packet level Fountain code
Fountain code
In coding theory, fountain codes are a class of erasure codes with the property that a potentially limitless sequence of encoding symbols can be generated from a given set of source symbols such that the original source symbols can ideally be recovered from any subset of the encoding symbols of...
s 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 Extended Report (RFC 3611), SIP
Session Initiation Protocol
The Session Initiation Protocol is an IETF-defined signaling protocol widely used for controlling communication sessions such as voice and video calls over Internet Protocol . The protocol can be used for creating, modifying and terminating two-party or multiparty sessions...
RTCP Summary Reports, H.460.9 Annex B (for H.323
H.323
H.323 is a recommendation from the ITU Telecommunication Standardization Sector that defines the protocols to provide audio-visual communication sessions on any packet network...
), H.248.30
Media Gateway Control Protocol (Megaco)
Megaco is a gateway control protocol. and an implementation of the Media Gateway Control Protocol architecture for controlling media gateways in Internet Protocol networks and the public switched telephone network...
and MGCP
Media Gateway Control Protocol (MGCP)
MGCP is an implementation of the Media Gateway Control Protocol architecture for controlling media gateways on Internet Protocol networks and the public switched telephone network . The general base architecture and programming interface is described in RFC 2805 and the current specific MGCP...
extensions. The RFC 3611 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 (because of jitter), packet loss/discard burst metrics (burst length/density, gap length/density), network delay, end system delay, signal / noise / echo level, Mean Opinion Score
Mean Opinion Score
The Mean Opinion Score test has been used for decades in telephony networks to obtain the human user's view of the quality of the network. In multimedia especially when codecs are used to compress the bandwidth requirement , the mean opinion score ...
s (MOS) and R factors and configuration information related to the jitter buffer.
RFC 3611 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. RFC 3611 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.
Layer-2 quality of service
A number of protocols that deal with the data link layerData link layer
The data link layer is layer 2 of the seven-layer OSI model of computer networking. It corresponds to, or is part of the link layer of the TCP/IP reference model....
and physical layer
Physical layer
The physical layer or layer 1 is the first and lowest layer in the seven-layer OSI model of computer networking. The implementation of this layer is often termed PHY....
include quality-of-service mechanisms that can be used to ensure that applications like VoIP work well even in congested scenarios. Some examples include:
- IEEE 802.11eIEEE 802.11eIEEE 802.11e-2005 or 802.11e is an approved amendment to the IEEE 802.11 standard that defines a set of Quality of Service enhancements for wireless LAN applications through modifications to the Media Access Control layer. The standard is considered of critical importance for delay-sensitive...
is an approved amendment to the IEEE 802.11IEEE 802.11IEEE 802.11 is a set of standards for implementing wireless local area network computer communication in the 2.4, 3.6 and 5 GHz frequency bands. They are created and maintained by the IEEE LAN/MAN Standards Committee . The base version of the standard IEEE 802.11-2007 has had subsequent...
standard that defines a set of quality-of-service enhancements for wireless LAN applications through modifications to the Media Access ControlMedia Access ControlThe media access control data communication protocol sub-layer, also known as the medium access control, is a sublayer of the data link layer specified in the seven-layer OSI model , and in the four-layer TCP/IP model...
(MAC) layer. The standard is considered of critical importance for delay-sensitive applications, such as Voice over Wireless IP. - IEEE 802.1pIEEE 802.1pIEEE P802.1p is the name of a task group active during 1995–98 responsible for adding traffic class expediting and dynamic multicast filtering to the IEEE 802.1D standard. Essentially, they provided a mechanism for implementing Quality of Service at the Media Access Control level...
defines 8 different classes of service (including one dedicated to voice) for traffic on layer-2 wired EthernetEthernetEthernet is a family of computer networking technologies for local area networks commercially introduced in 1980. Standardized in IEEE 802.3, Ethernet has largely replaced competing wired LAN technologies....
. - The ITU-TITU-TThe ITU Telecommunication Standardization Sector is one of the three sectors of the International Telecommunication Union ; it coordinates standards for telecommunications....
G.hnG.hnG.hn is the common name for a home network technology family of standards developed under the International Telecommunication Union's Standardization arm and promoted by the HomeGrid Forum...
standard, which provides a way to create a high-speed (up to 1 gigabit per second) Local area networkLocal area networkA local area network is a computer network that interconnects computers in a limited area such as a home, school, computer laboratory, or office building...
using existing home wiring (power linesPower line communicationPower line communication or power line carrier , also known as power line digital subscriber line , mains communication, power line telecom , power line networking , or broadband over power lines are systems for carrying data on a conductor also used for electric power transmission.A wide range...
, phone lines and coaxial cablesEthernet over coaxEthernet over Coax is a family of technologies that supports the transmission of Ethernet frames over coaxial cable.- History :The first Ethernet standard, known as 10BASE5 in the family of IEEE 802.3, specified baseband operation over coaxial cable...
). G.hn provides QoS by means of "Contention-Free Transmission Opportunities" (CFTXOPs) which are allocated to flows (such as a VoIP call) which require QoS and which have negotiated a "contract" with the network controllers.
Susceptibility to power failure
Telephones for traditional residential analog service are usually connected directly to telephone company phone lines which provide direct current to power most basic analog handsets independently of locally available power.IP Phone
IP Phone
A VoIP phone uses voice over IP technologies allowing telephone calls to be made over an IP network such as the Internet instead of the ordinary PSTN system. Calls can traverse the Internet, or a private IP network such as that of a company...
s and VoIP telephone adapters connect to routers or cable modem
Cable modem
A cable modem is a type of network bridge and modem that provides bi-directional data communication via radio frequency channels on a HFC and RFoG infrastructure. Cable modems are primarily used to deliver broadband Internet access in the form of cable Internet, taking advantage of the high...
s which typically depend on the availability of mains electricity
Mains electricity
Mains is the general-purpose alternating current electric power supply. In the US, electric power is referred to by several names including household power, household electricity, powerline, domestic power, wall power, line power, AC power, city power, street power, and grid power...
or locally generated power. Some VoIP service providers use customer premise equipment (e.g., cablemodems) with battery-backed power supplies to assure uninterrupted service for up to several hours in case of local power failures. Such battery-backed devices typically are designed for use with analog handsets.
Some VoIP service providers implement services to route calls to other telephone services of the subscriber, such a cellular phone, in the event that the customer's network device is inaccessible to terminate the call.
The susceptibility of phone service to power failures is a common problem even with traditional analog service in areas where many customers purchase modern telephone units that operate with wireless handsets to a base station, or that have other modern phone features, such as built-in voicemail or phone book features.
Emergency calls
The nature of IPInternet Protocol
The Internet Protocol is the principal communications protocol used for relaying datagrams across an internetwork using the Internet Protocol Suite...
makes it difficult to locate network users geographically. Emergency call
Emergency telephone number
Many countries' public telephone networks have a single emergency telephone number, sometimes known as the universal emergency telephone number or occasionally the emergency services number, that allows a caller to contact local emergency services for assistance. The emergency telephone number may...
s, therefore, cannot easily be routed to a nearby call center. Sometimes, VoIP systems may route emergency calls to a non-emergency phone line at the intended department; in the United States, at least one major police department has strongly objected to this practice as potentially endangering the public.
A fixed line phone has a direct relationship between a telephone number and a physical location. If an emergency call comes from that number, then the physical location is known.
In the IP world, it is not so simple. A broadband provider may know the location where the wires terminate, but this does not necessarily allow the mapping of an IP address to that location. IP addresses are often dynamically assigned, so the ISP may allocate an address for online access, or at the time a broadband router is engaged. The ISP recognizes individual IP addresses, but does not necessarily know to which physical location it corresponds. The broadband service provider knows the physical location, but is not necessarily tracking the IP addresses in use.
There are more complications since IP allows a great deal of mobility. For example, a broadband connection can be used to dial a virtual private network that is employer-owned. When this is done, the IP address being used will belong to the range of the employer, rather than the address of the ISP, so this could be many kilometres away or even in another country. To provide another example: if mobile data is used, e.g., a 3G mobile handset or USB wireless broadband adapter, then the IP address has no relationship with any physical location, since a mobile user could be anywhere that there is network coverage, even roaming via another cellular company.
In short, there is no relationship between IP address and physical location, so the address itself reveals no useful information for the emergency services.
At the VoIP level, a phone or gateway may identify itself with a SIP
Session Initiation Protocol
The Session Initiation Protocol is an IETF-defined signaling protocol widely used for controlling communication sessions such as voice and video calls over Internet Protocol . The protocol can be used for creating, modifying and terminating two-party or multiparty sessions...
registrar by using a username and password. So in this case, the Internet Telephony Service Provider (ITSP) knows that a particular user is online, and can relate a specific telephone number to the user. However, it does not recognize how that IP traffic was engaged. Since the IP address itself does not necessarily provide location information presently, today a "best efforts" approach is to use an available database to find that user and the physical address the user chose to associate with that telephone number—clearly an imperfect solution.
VoIP Enhanced 911
Enhanced 911
Enhanced 911, E-911 or E911 in North America is one example of the modern evolution of telecommunications based system meant as an easy way to link people experiencing an emergency with the public resources that can help. The dial-three-digits concept first originated in the United Kingdom in 1937....
(E911) is a method by which VoIP providers in the United States 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 VoIP providers that provide access to the public switched telephone network are required to implement E911, a service for which the subscriber may be charged. Participation in E911 is not required and customers may opt-out of E911 service.
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
Assisted GPS
Assisted GPS, generally abbreviated as A-GPS or aGPS, is a system which can, under certain conditions, improve the startup performance, or time-to-first-fix of a GPS satellite-based positioning system. It is used extensively with GPS-capable cellular phones as its development was accelerated by...
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.
Lack of redundancy
With the current separation of the Internet and the PSTN, a certain amount of redundancy is provided. An Internet outage does not necessarily mean that a voice communication outage will occur simultaneously, allowing individuals to call for emergency services and many businesses to continue to operate normally. In situations where telephone services become completely reliant on the Internet infrastructure, a single-point failure can isolate communities from all communication, including Enhanced 911Enhanced 911
Enhanced 911, E-911 or E911 in North America is one example of the modern evolution of telecommunications based system meant as an easy way to link people experiencing an emergency with the public resources that can help. The dial-three-digits concept first originated in the United Kingdom in 1937....
and equivalent services in other locales. However, the internet as designed by DARPA in the early 1980s was specifically designed to be fault tolerant under adverse conditions. Even during the 9/11 attacks on the World Trade Centers the internet routed data around the failed nodes that were housed in or near the towers. So single point failures while possible in some geographic areas are not the norm for the internet as a whole.
Number portability
Local number portabilityLocal number portability
Local number portability for fixed lines, and full mobile number portability for mobile phone lines, refers to the ability to transfer either an existing fixed-line or mobile telephone number assigned by a local exchange carrier and reassign it to another carrier...
(LNP) and Mobile number portability
Mobile number portability
Mobile number portability enables mobile telephone users to retain their mobile telephone numbers when changing from one mobile network operator to another.- General overview :MNP is implemented in different ways across the globe...
(MNP) also impact VoIP business. In November 2007, the Federal Communications Commission
Federal Communications Commission
The Federal Communications Commission is an independent agency of the United States government, created, Congressional statute , and with the majority of its commissioners appointed by the current President. The FCC works towards six goals in the areas of broadband, competition, the spectrum, the...
in the United States released an order extending number portability obligations to interconnected VoIP providers and carriers that support VoIP providers. Number portability is a service that allows a subscriber to select a new telephone carrier without requiring a new number to be issued. Typically, it is the responsibility of the former carrier to "map" the old number to the undisclosed number assigned by the new carrier. This is achieved by maintaining a database of numbers. A dialed number is initially received by the original carrier and quickly rerouted to the new carrier. Multiple porting references must be maintained even if the subscriber returns to the original carrier. The FCC mandates carrier compliance with these consumer-protection stipulations.
A voice call originating in the VoIP environment also faces challenges to reach its destination if the number is routed to a mobile phone number on a traditional mobile carrier. VoIP has been identified in the past as a Least Cost Routing
Least cost routing
In voice telecommunications, least-cost routing is the process of selecting the path of outbound communications traffic based on cost. Within a telecoms carrier, an LCR team might periodically choose between routes from several or even hundreds of carriers for destinations across the world...
(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. This rating is subject to some debate given the complexity of call routing created by number portability. 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 must now determine 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 which home network a mobile phone number belongs to. As the popularity of VoIP increases in the enterprise markets because of least cost routing
Least cost routing
In voice telecommunications, least-cost routing is the process of selecting the path of outbound communications traffic based on cost. Within a telecoms carrier, an LCR team might periodically choose between routes from several or even hundreds of carriers for destinations across the world...
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 by assuring that the voice call will actually work, VoIP service providers are able to offer business subscribers the level of reliability they require.
PSTN integration
E.164E.164
E.164 is an ITU-T recommendation that defines the international public telecommunication numbering plan used in the PSTN and some other data networks. It also defines the format of telephone numbers. E.164 numbers can have a maximum of fifteen digits and are usually written with a + prefix...
is a global FGFnumbering standard for both the PSTN
Public switched telephone network
The public switched telephone network is the network of the world's public circuit-switched telephone networks. It consists of telephone lines, fiber optic cables, microwave transmission links, cellular networks, communications satellites, and undersea telephone cables, all inter-connected by...
and PLMN
Public land mobile network
A public land mobile network is a regulatory term in telecommunications. A PLMN is a network that is established and operated by an administration or by a recognized operating agency for the specific purpose of providing land mobile telecommunications services to the public.A PLMN is identified...
. Most VoIP implementations support E.164
E.164
E.164 is an ITU-T recommendation that defines the international public telecommunication numbering plan used in the PSTN and some other data networks. It also defines the format of telephone numbers. E.164 numbers can have a maximum of fifteen digits and are usually written with a + prefix...
to allow calls to be routed to and from VoIP subscribers and the PSTN/PLMN. VoIP implementations can also allow other identification techniques to be used. For example, Skype allows subscribers to choose "Skype names" (usernames) whereas SIP implementations can use URI
Uniform Resource Identifier
In computing, a uniform resource identifier is a string of characters used to identify a name or a resource on the Internet. Such identification enables interaction with representations of the resource over a network using specific protocols...
s similar to email addresses
E-mail address
An email address identifies an email box to which email messages are delivered. An example format of an email address is lewis@example.net which is read as lewis at example dot net...
. Often VoIP implementations employ methods of translating non-E.164 identifiers to E.164 numbers and vice-versa, such as the Skype-In service provided by Skype and the ENUM
Telephone Number Mapping
Telephone number mapping is the process of unifying the telephone number system of the public switched telephone network with the Internet addressing and identification name spaces. Telephone numbers are systematically organized in the E.164 standard, while the Internet uses the Domain Name System...
service in IMS and SIP.
Echo can also be an issue for PSTN integration. Common causes of echo include impedance mismatches
Impedance matching
In electronics, impedance matching is the practice of designing the input impedance of an electrical load to maximize the power transfer and/or minimize reflections from the load....
in analog circuitry and acoustic coupling of the transmit and receive signal at the receiving end.
Security
VoIP telephone systems are susceptible to attacks as are any internet-connected devices. This means that hackerHacker (computer security)
In computer security and everyday language, a hacker is someone who breaks into computers and computer networks. Hackers may be motivated by a multitude of reasons, including profit, protest, or because of the challenge...
s who know about these vulnerabilities (such as insecure passwords) can institute denial-of-service attacks, harvest customer data, record conversations and break into voice mailboxes.
Another challenge is routing VoIP traffic through firewalls and network address translators. Private Session Border Controller
Session Border Controller
A session border controller is a device regularly deployed in Voice over Internet Protocol networks to exert control over the signaling and usually also the media streams involved in setting up, conducting, and tearing down telephone calls or other interactive media communications.SBC's original...
s are used along with firewalls to enable VoIP calls to and from protected networks. For example, 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 NATs involve using protocols such as STUN
STUN
STUN is a standardized set of methods, including a network protocol, used in NAT traversal for applications of real-time voice, video, messaging, and other interactive IP communications....
or Interactive Connectivity Establishment
Interactive Connectivity Establishment
Interactive Connectivity Establishment is a technique used in computer networking involving network address translators in Internet applications of Voice over Internet Protocol , peer-to-peer communications, video, instant messaging and other interactive media...
(ICE).
Many consumer VoIP solutions do not support encryption, although having a secure phone is much easier to implement with VoIP than traditional phone lines. As a result, it is relatively easy to eavesdrop on VoIP calls and even change their content. An attacker with a packet sniffer could intercept your VoIP calls if you are not on a secure VLAN. However, physical security of the switches within an enterprise and the facility security provided by ISPs make packet capture less of a problem than originally foreseen. Further research has shown that tapping into a fiber optic network without detection is difficult if not impossible. This means that once a voice packet is within the internet backbone it is relatively safe from interception.
There are open source solutions, such as Wireshark
Wireshark
Wireshark is a free and open-source packet analyzer. It is used for network troubleshooting, analysis, software and communications protocol development, and education...
, that facilitate sniffing of VoIP conversations. A modicum of security is afforded by patented audio codecs in proprietary implementations that are not easily available for open source applications; however, such security through obscurity
Security through obscurity
Security through obscurity is a pejorative referring to a principle in security engineering, which attempts to use secrecy of design or implementation to provide security...
has not proven effective in other fields. Some vendors also use compression, which may make eavesdropping
Eavesdropping
Eavesdropping is the act of secretly listening to the private conversation of others without their consent, as defined by Black's Law Dictionary...
more difficult. However, real security requires encryption and cryptographic authentication which are not widely supported at a consumer level. The existing security standard Secure Real-time Transport Protocol
Secure Real-time Transport Protocol
The Secure Real-time Transport Protocol defines a profile of RTP , intended to provide encryption, message authentication and integrity, and replay protection to the RTP data in both unicast and multicast applications...
(SRTP) and the new ZRTP
ZRTP
ZRTP is a cryptographic key-agreement protocol to negotiate the keys for encryption between two end points in a Voice over Internet Protocol phone telephony call based on the Real-time Transport Protocol. It uses Diffie-Hellman key exchange and the Secure Real-time Transport Protocol for...
protocol are available on Analog Telephone Adapters (ATAs)
Analog telephony adapter
An analog telephony adapter, or analog telephone adapter, is a device used to connect one or more standard analog telephones to a digital telephone system or a non-standard telephone system....
as well as various softphone
Softphone
A softphone is a software program for making telephone calls over the Internet using a general purpose computer, rather than using dedicated hardware. Often a softphone is designed to behave like a traditional telephone, sometimes appearing as an image of a phone, with a display panel and buttons...
s. It is possible to use IPsec
IPsec
Internet Protocol Security is a protocol suite for securing Internet Protocol communications by authenticating and encrypting each IP packet of a communication session...
to secure P2P VoIP by using opportunistic encryption
Opportunistic encryption
Opportunistic Encryption refers to any system that, when connecting to another system, attempts to encrypt the communications channel otherwise falling back to unencrypted communications. This method requires no pre-arrangement between the two systems.Opportunistic encryption can be used to...
. Skype does not use SRTP, but uses encryption which is transparent to the Skype provider. In 2005, Skype invited a researcher, Dr Tom Berson, to assess the security of the Skype software, and his conclusions are available in a published report.
Securing VoIP
To prevent the above security concerns government and military organizations are using Voice over Secure IP (VoSIP), Secure Voice over IP (SVoIP), and Secure Voice over Secure IP (SVoSIP) to protect confidential and classified VoIP communications. Secure Voice over IP is accomplished by encrypting VoIP with Type 1 encryptionType 1 encryption
In cryptography, a Type 1 product is a device or system certified by the National Security Agency for use in cryptographically securing classified U.S...
. Secure Voice over Secure IP is accomplished by using Type 1 encryption
Type 1 encryption
In cryptography, a Type 1 product is a device or system certified by the National Security Agency for use in cryptographically securing classified U.S...
on a classified network, like SIPRNet
SIPRNet
The Secret Internet Protocol Router Network is "a system of interconnected computer networks used by the United States Department of Defense and the U.S. Department of State to transmit classified information by packet switching over the TCP/IP protocols in a 'completely secure' environment"...
. Public Secure VoIP is also available with free GNU programs.
Caller ID
Caller IDCaller ID
Caller ID , also called calling line identification or calling number identification or Calling Line Identification Presentation , is a telephone service, available in analog and digital phone systems and most Voice over Internet Protocol applications, that transmits a caller's number to...
support among VoIP providers varies, but is provided by the majority of VoIP providers.
Many VoIP carriers allow callers to configure arbitrary Caller ID information, thus permitting spoofing attack
Spoofing attack
In the context of network security, a spoofing attack is a situation in which one person or program successfully masquerades as another by falsifying data and thereby gaining an illegitimate advantage.- Spoofing and TCP/IP :...
s. Business grade VoIP equipment and software often makes it easy to modify caller ID information, providing many businesses great flexibility.
The Truth in Caller ID Act has been in preparation in the US Congress since 2006, but as of January 2009 still has not been enacted. This bill proposes to make it a crime in the United States to "knowingly transmit misleading or inaccurate caller identification information with the intent to defraud, cause harm, or wrongfully obtain anything of value ..."
Compatibility with traditional analog telephone sets
Some analog telephone adapters do not decode pulse dialing from older phones. They may only work with push-button telephonePush-button telephone
The push-button telephone was first invented in 1941, and is a telephone with push-buttons or keys, and which eventually replaced rotary dial telephones that were first used in 1891. The first push-button telephone was invented in the labs of Bell Telephone; however, these models were only...
s using the touch-tone system. The VoIP user may use a pulse-to-tone converter, if needed.
Fax handling
Support for sending faxes over VoIP implementations is still limited. The existing voice codecCodec
A codec is a device or computer program capable of encoding or decoding a digital data stream or signal. The word codec is a portmanteau of "compressor-decompressor" or, more commonly, "coder-decoder"...
s are not designed for fax transmission; they are designed to digitize an analog representation of a human voice efficiently. However, the inefficiency of digitizing an analog representation (modem signal) of a digital representation (a document image) of analog data (an original document) more than negates any bandwidth advantage of VoIP. In other words, the fax "sounds" simply do not fit in the VoIP channel. An alternative IP-based solution for delivering fax-over-IP called T.38
T.38
T.38 is an ITU recommendation for allowing transmission of fax over IP networks in real time.- History :The T.38 fax relay standard was devised in 1998 as a way to permit faxes to be transported across IP networks between existing Group 3 fax terminals. T.4 and related fax standards were published...
is available. Sending faxes using VoIP is sometimes referred to as FoIP, or Fax over IP.
The T.38 protocol is designed to compensate for the differences between traditional packet-less communications over analog lines and packet based transmissions which are the basis for IP communications. 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. TCP is better suited for use between two IP devices. However, older fax machines, connected to an analog system, benefit from UDP near real-time characteristics due to the "no recovery rule" when a UDP packet is lost or an error occurs during transmission. UDP transmissions are preferred as they do not require testing for dropped packets and as such since each T.38 packet transmission includes a majority of the data sent in the prior packet, a T.38 termination point has a higher degree of success in re-assembling the fax transmission back into its original form for interpretation by the end device. This in an attempt to overcome the obstacles of simulating real time transmissions using packet based protocol.
There have been updated versions of T.30 to resolve the fax over IP issues, which is the core fax protocol. Some newer high end fax machines have T.38 built-in capabilities which allow the user to plug right into the network and transmit/receive faxes in native T.38 like the Ricoh 4410NF Fax Machine. A unique feature of T.38 is that each packet contains a portion of the main data sent in the previous packet. With T.38, two successive lost packets are needed 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 an increased likelihood that you will receive enough of the transmission to satisfy the requirements of the fax machine for output of the sent document.
Support for other telephony devices
Another challenge for VoIP implementations is the proper handling of outgoing calls from other telephony devices such as digital video recorderDigital video recorder
A digital video recorder , sometimes referred to by the merchandising term personal video recorder , is a consumer electronics device or application software that records video in a digital format to a disk drive, USB flash drive, SD memory card or other local or networked mass storage device...
s, satellite television
Satellite television
Satellite television is television programming delivered by the means of communications satellite and received by an outdoor antenna, usually a parabolic mirror generally referred to as a satellite dish, and as far as household usage is concerned, a satellite receiver either in the form of an...
receivers, alarm
Alarm
An alarm device or system of alarm devices gives an audible or visual alarm signal about a problem or condition.Alarm devices include:* burglar alarms, designed to warn of burglaries; this is often a silent alarm: the police or guards are warned without indication to the burglar, which increases...
systems, conventional modem
Modem
A modem is a device that modulates an analog carrier signal to encode digital information, and also demodulates such a carrier signal to decode the transmitted information. The goal is to produce a signal that can be transmitted easily and decoded to reproduce the original digital data...
s and other similar devices that depend on access to a PSTN telephone
Telephone
The telephone , colloquially referred to as a phone, is a telecommunications device that transmits and receives sounds, usually the human voice. Telephones are a point-to-point communication system whose most basic function is to allow two people separated by large distances to talk to each other...
line for some or all of their functionality.
These types of calls sometimes complete without any problems, but in other cases they fail. If VoIP and cellular
Cellular network
A cellular network is a radio network distributed over land areas called cells, each served by at least one fixed-location transceiver known as a cell site or base station. When joined together these cells provide radio coverage over a wide geographic area...
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 PSTN telephone line would be available in consumer's homes.
Legal issues
As the popularity of VoIP grows, governments are becoming more interested in regulating VoIP in a manner similar to PSTN services.Another legal issue that the US Congress is debating concerns changes to the Foreign Intelligence Surveillance Act. The issue in question is calls between Americans and foreigners. The National Security Agency (NSA) is not authorized to tap Americans' conversations without a warrant—but the Internet, and specifically VoIP does not draw as clear a line to the location of a caller or a call's recipient as the traditional phone system does. As VoIP's low cost and flexibility convinces more and more organizations to adopt the technology, the surveillance for law enforcement agencies becomes more difficult. VoIP technology has also increased security concerns because VoIP and similar technologies have made it more difficult for the government to determine where a target is physically located when communications are being intercepted, and that creates a whole set of new legal challenges.
In the US, the Federal Communications Commission
Federal Communications Commission
The Federal Communications Commission is an independent agency of the United States government, created, Congressional statute , and with the majority of its commissioners appointed by the current President. The FCC works towards six goals in the areas of broadband, competition, the spectrum, the...
now requires all interconnected VoIP service providers to comply with requirements comparable to those for traditional telecommunications service providers. VoIP operators in the US are required to support local number portability
Local number portability
Local number portability for fixed lines, and full mobile number portability for mobile phone lines, refers to the ability to transfer either an existing fixed-line or mobile telephone number assigned by a local exchange carrier and reassign it to another carrier...
; make service accessible to people with disabilities; pay regulatory fees, universal service
Universal service
Universal service is an economic, legal and business term used mostly in regulated industries, referring to the practice of providing a baseline level of services to every resident of a country...
contributions, and other mandated payments; and enable law enforcement authorities to conduct surveillance pursuant to the Communications Assistance for Law Enforcement Act
Communications Assistance for Law Enforcement Act
The Communications Assistance for Law Enforcement Act is a United States wiretapping law passed in 1994, during the presidency of Bill Clinton...
(CALEA). "Interconnected" VoIP operators also must provide Enhanced 911
Enhanced 911
Enhanced 911, E-911 or E911 in North America is one example of the modern evolution of telecommunications based system meant as an easy way to link people experiencing an emergency with the public resources that can help. The dial-three-digits concept first originated in the United Kingdom in 1937....
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 US telecommunications regulations, including an entitlement to interconnection
Interconnection
In telecommunications, interconnection is the physical linking of a carrier's network with equipment or facilities not belonging to that network...
and exchange of traffic with incumbent local exchange carrier
Incumbent local exchange carrier
An ILEC, short for incumbent local exchange carrier, is a local telephone company in the United States that was in existence at the time of the breakup of AT&T into the Regional Bell Operating Companies , also known as the "Baby Bells." The ILEC is the former Bell System or Independent Telephone...
s 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
Panama
Panama , officially the Republic of Panama , is the southernmost country of Central America. Situated on the isthmus connecting North and South America, it is bordered by Costa Rica to the northwest, Colombia to the southeast, the Caribbean Sea to the north and the Pacific Ocean to the south. The...
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
Ethiopia
Ethiopia , officially known as the Federal Democratic Republic of Ethiopia, is a country located in the Horn of Africa. It is the second-most populous nation in Africa, with over 82 million inhabitants, and the tenth-largest by area, occupying 1,100,000 km2...
, 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 the popularity of VoIP reduced the income generated by the state owned telecommunication company.
In the European Union
European Union
The European Union is an economic and political union of 27 independent member states which are located primarily in Europe. The EU traces its origins from the European Coal and Steel Community and the European Economic Community , formed by six countries in 1958...
, 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
Plain old telephone service
Plain old telephone service is the voice-grade telephone service that remains the basic form of residential and small business service connection to the telephone network in many parts of the world....
call should not be inside India.
In the UAE
United Arab Emirates
The United Arab Emirates, abbreviated as the UAE, or shortened to "the Emirates", is a state situated in the southeast of the Arabian Peninsula in Western Asia on the Persian Gulf, bordering Oman, and Saudi Arabia, and sharing sea borders with Iraq, Kuwait, Bahrain, Qatar, and Iran.The UAE is a...
and Oman it is illegal to use any form of VoIP, to the extent that Web sites of Skype
Skype
Skype is a software application that allows users to make voice and video calls and chat over the Internet. Calls to other users within the Skype service are free, while calls to both traditional landline telephones and mobile phones can be made for a fee using a debit-based user account system...
and Gizmo5
Gizmo5
Gizmo5 was a Voice over Internet Protocol communications network and a proprietary freeware soft phone for that network. On November 12, 2009, Google announced that it had acquired Gizmo5...
are blocked. Providing or using VoIP services is illegal in Oman. Those who violate the law stand to be fined 50,000 Omani Rial (about 130,317 US dollars) or spend two years in jail or both. In 2009, police in Oman have raided 121 internet cafes throughout the country and arrested 212 people for using/providing VoIP services.
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 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
United States Forces Korea
United States Forces Korea refers to the ground, air and naval divisions of the United States armed forces stationed in South Korea....
members residing on USFK bases threatened to block off access to VoIP services used by USFK members 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 US-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 US VoIP providers.
Historical milestones
- 1974 – The Institute of Electrical and Electronic Engineers (IEEE) published a paper titled "A Protocol for Packet Network Interconnection."
- 1981 – IPv4IPv4Internet Protocol version 4 is the fourth revision in the development of the Internet Protocol and the first version of the protocol to be widely deployed. Together with IPv6, it is at the core of standards-based internetworking methods of the Internet...
is described in RFC 791. - 1985 – The National Science FoundationNational Science FoundationThe National Science Foundation is a United States government agency that supports fundamental research and education in all the non-medical fields of science and engineering. Its medical counterpart is the National Institutes of Health...
commissions the creation of NSFNETNational Science Foundation NetworkThe National Science Foundation Network was a program of coordinated, evolving projects sponsored by the National Science Foundation beginning in 1985 to promote advanced research and education networking in the United States...
. - 1994 - Alon CohenAlon CohenAlon Cohen is the co-founder of VocalTec Inc. and the inventor of the Audio Transceiver that enabled the creation of Voice Over Networks products and eventually the VoIP industry, that changed the face of the telecommunication industry...
and Lior HaramatyLior HaramatyLior Haramaty is the co-founder of VocalTec Inc. and the inventor of the Audio Transceiver that enabled the creation of Voice Over Networks products and eventually the VoIP industry, that changed the face of the telecommunication industry.VocalTec is recognized as the first company to provide...
invent Voice over IP. - 1995 – VocalTecVocalTecVocalTec Communications Inc. , is an Israeli telecom equipment provider. The company was founded in 1989 by Alon Cohen and Lior Haramaty, who invented and patented the first Voice over IP audio transceiver...
releases the first commercial Internet phone software.- Beginning in 1995, Intel, MicrosoftMicrosoftMicrosoft Corporation is an American public multinational corporation headquartered in Redmond, Washington, USA that develops, manufactures, licenses, and supports a wide range of products and services predominantly related to computing through its various product divisions...
and Radvision initiated standardization activities for VoIP communications system.
- Beginning in 1995, Intel, Microsoft
- 1996 –
- ITU-TITU-TThe ITU Telecommunication Standardization Sector is one of the three sectors of the International Telecommunication Union ; it coordinates standards for telecommunications....
begins development of standards for the transmission and signaling of voice communications over Internet Protocol networks with the H.323H.323H.323 is a recommendation from the ITU Telecommunication Standardization Sector that defines the protocols to provide audio-visual communication sessions on any packet network...
standard. - US telecommunication companies petition the US Congress to ban Internet phone technology.
- ITU-T
- 1997 – Level 3Level 3 CommunicationsLevel 3 Communications is a telecommunications and Internet service provider headquartered in Broomfield, Colorado.It operates a Tier 1 network. The company provides core transport, IP, voice, video and content delivery for most of the medium to large Internet carriers in North America and Europe...
began development of its first softswitchSoftswitchA softswitch is a central device in a telecommunications network which connects telephone calls from one phone line to another, typically via the internet, entirely by means of software running on a general-purpose computer system...
, a term they coined in 1998. - 1999 –
- The Session Initiation ProtocolSession Initiation ProtocolThe Session Initiation Protocol is an IETF-defined signaling protocol widely used for controlling communication sessions such as voice and video calls over Internet Protocol . The protocol can be used for creating, modifying and terminating two-party or multiparty sessions...
(SIP) specification RFC 2543 is released. - Mark Spencer of DigiumDigiumDigium, Inc. is a privately held communications technology company based in Huntsville, Alabama. Digium specializes in developing and manufacturing communications hardware and telephony software, most notably the open-source telephony platform Asterisk....
develops the first open source private branch exchange (PBX) software (AsteriskAsterisk (PBX)Asterisk is a software implementation of a telephone private branch exchange ; it was created in 1999 by Mark Spencer of Digium. Like any PBX, it allows attached telephones to make calls to one another, and to connect to other telephone services including the public switched telephone network and...
).
- The Session Initiation Protocol
- 2004 – Commercial VoIP service providers proliferate.
Pronunciation
The acronym VoIP has been pronounced variably since the inception of the term. Apart from spelling out the acronym letter by letter, vē'ō'ī'pē (vee-oh-eye-pee), there are three likely possible pronunciations: vō'ī'pē (vo-eye-pee) and vō'ip (vo-ipp), have been used, but generally, the single syllable vŏy'p (voyp, as in voice) may be the most common within the industry.See also
- Audio over IPAudio over IPStreaming audio over IP networks is being increasingly used by broadcasting companies, among others, to provide high-quality audio feeds over distance across an IP network such as the Internet. The application is also known as audio contribution over IP in reference to the programming...
- Capillary routingCapillary routingIn networking and in graph theory, capillary routing, for a given network, is a multi-path solution between a pair of source and destination nodes...
- Call shopCall shopA call shop is a business that provides physical phones for the purpose of making long-distance telephone calls. There are two main types of call shop payment methods; prepaid and postpaid.- Call shop requirements :...
- Click to talk
- Communications Assistance For Law Enforcement ActCommunications Assistance for Law Enforcement ActThe Communications Assistance for Law Enforcement Act is a United States wiretapping law passed in 1994, during the presidency of Bill Clinton...
- Comparison of VoIP softwareComparison of VoIP softwareVoIP software is used to conduct telephone-like voice conversations across Internet Protocol based networks. VoIP stands for "Voice over IP". For residential markets, VoIP phone service is often cheaper than traditional public switched telephone network service and can remove geographic...
- Computer conferencingComputer conferencingIn telecommunication, the term computer conferencing has the following meanings:#Teleconferencing supported by one or more computers.#An arrangement in which access, by multiple users, to a common database is mediated by a controlling computer....
- Dial peerDial peerA dial peer, also termed addressable call endpoint, initiates or obtains calls within a telephone network.-See also:*VOIP*Session Initiation Protocol*Foreign exchange station*Foreign exchange office*Off Premise Extension*PLAR-External sources:...
- Differentiated servicesDifferentiated servicesDifferentiated Services or DiffServ is a computer networking architecture that specifies a simple, scalable and coarse-grained mechanism for classifying and managing network traffic and providing Quality of Service on modern IP networks...
- High bit rate audio video over Internet ProtocolHigh bit rate audio video over Internet ProtocolHigh bit rate media transport formerly known as High bit rate audio video over IP , is a proposed standard for data encapsulation and forward error correction of high bit rate contribution oriented video/audio feed services, up to 3 Gbit/s over Ethernet networks. HBRMT is being developed by...
- Integrated servicesIntegrated servicesIn computer networking, IntServ or integrated services is an architecture that specifies the elements to guarantee quality of service on networks...
- Internet faxInternet faxInternet fax uses the Internet to receive and send faxes.Internet faxing, "e-Fax" or "online faxing" is a general term which refers to sending a document facsimile using the Internet, rather than using only phone networks with a fax machine.Depending on the specific method/implementation ,...
- IP Multimedia SubsystemIP Multimedia SubsystemThe IP Multimedia Subsystem or IP Multimedia Core Network Subsystem is an architectural framework for delivering Internet Protocol multimedia services. It was originally designed by the wireless standards body 3rd Generation Partnership Project , as a part of the vision for evolving mobile...
- IP PhoneIP PhoneA VoIP phone uses voice over IP technologies allowing telephone calls to be made over an IP network such as the Internet instead of the ordinary PSTN system. Calls can traverse the Internet, or a private IP network such as that of a company...
- Managed Facility Voice NetworkManaged Facility Voice NetworkA managed facilities-based voice network, or MFVN, is a physical network owned and operated by a voice service provider that delivers traditional telephone service via a loop start analog telephone interface. MFVNs are interconnected with the public switched telephone network and provide dialtone...
- Mobile VoIPMobile VoIPMobile VoIP or simply mVoIP is an extension of mobility to a Voice over IP network. Two types of communication are generally supported: cordless/DECT/PCS protocols for short range or campus communications where all base stations are linked into the same LAN, and wider area communications using...
- Mouth-to-ear delay
- Network Voice ProtocolNetwork Voice ProtocolThe Network Voice Protocol was a pioneering computer network protocol for transporting human speech over packetized communications networks...
- Predictive dialers
- Publicly Available Telephone ServicesPublicly Available Telephone ServicesPublicly Available Telephone Services or PATS means a service available to the public for originating and receiving national and international calls and access to emergency services through a telephone number or numbers in a national or international telephone numbering plan, and may, where...
- Push-button telephonePush-button telephoneThe push-button telephone was first invented in 1941, and is a telephone with push-buttons or keys, and which eventually replaced rotary dial telephones that were first used in 1891. The first push-button telephone was invented in the labs of Bell Telephone; however, these models were only...
- RTP audio video profile
- Secure telephoneSecure telephoneA secure telephone is a telephone that provides voice security in the form of end-to-end encryption for the telephone call, and in some cases also the mutual authentication of the call parties, protecting them against a man-in-the-middle attack...
- SIP TrunkingSIP TrunkingSIP trunking is a Voice over Internet Protocol service based on the Session Initiation Protocol by which Internet telephony service providers deliver telephone services and unified communication to customers equipped with private branch exchange facilities.-Domains:The architecture of SIP...
- UNIStimUNIStimUNIStim is a Telecommunications protocol developed by Nortel for IP Phone and IP PBX communications....
- Voice VPN
- VoiceXMLVoiceXMLVoiceXML is the W3C's standard XML format for specifying interactive voice dialogues between a human and a computer. It allows voice applications to be developed and deployed in an analogous way to HTML for visual applications. Just as HTML documents are interpreted by a visual web browser,...
- VoIP recordingVoIP recordingVoice over Internet Protocol recording is a subset of telephone recording or voice logging, first used by call centers and now being used by all types of businesses...
- Web-based VoIP
VoIP services and providers
- HamSphereHamSphereHamSphere is a software Amateur Radio simulator that allows licensed radio amateurs and unlicensed enthusiasts to communicate with one another using a simulated ionosphere. It was designed by Kelly Lindman, a radio amateur with callsign 5B4AIT....
- OomaOomaOoma is a consumer telecommunications company based in Palo Alto, California, in the United States that allows its users to make phone calls anywhere in the United States with no monthly service fees. After an initial purchase, customers only pay applicable government taxes and access fees, around...
- RingCentralRingCentralRingCentral provides cloud computing based phone systems, designed to help small businesses manage mobile, fax, and email communications. Its products include RingCentral Office, RingCentral Mobile, and RingCentral Fax. - History :...
- Radvision
- SkypeSkypeSkype is a software application that allows users to make voice and video calls and chat over the Internet. Calls to other users within the Skype service are free, while calls to both traditional landline telephones and mobile phones can be made for a fee using a debit-based user account system...