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Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of technologies for the delivery of
voice communication Speech is a human vocal communication using language. Each language uses phonetic combinations of vowel and consonant sounds that form the sound of its words (that is, all English words sound different from all French words, even if they are th ...
s and
multimedia Multimedia is a form of communication that uses a combination of different content forms such as text, audio, images, animations, or video into a single interactive presentation, in contrast to tradition ...
sessions over
Internet Protocol The Internet Protocol (IP) is the network layer communications protocol in the Internet protocol suite for relaying datagrams across network boundaries. Its routing function enables internetworking, and essentially establishes the Internet. IP h ...
(IP) networks, such as the
Internet The Internet (or internet) is the global system of interconnected computer networks that uses the Internet protocol suite (TCP/IP) to communicate between networks and devices. It is a '' network of networks'' that consists of private, pub ...
. The terms Internet telephony, broadband telephony, and broadband phone service specifically refer to the provisioning of communications services (voice,
fax Fax (short for facsimile), sometimes called telecopying or telefax (the latter short for telefacsimile), is the telephonic transmission of scanned printed material (both text and images), normally to a telephone number connected to a printer o ...
,
SMS Short Message/Messaging Service, commonly abbreviated as SMS, is a text messaging service component of most telephone, Internet and mobile device systems. It uses standardized communication protocols that let mobile devices exchange short text ...
, voice-messaging) over the Internet, rather than via the
public switched telephone network The public switched telephone network (PSTN) provides Communications infrastructure, infrastructure and services for public Telecommunications, telecommunication. The PSTN is the aggregate of the world's circuit-switched telephone networks that ...
(PSTN), also known as
plain old telephone service Plain old telephone service (POTS), or plain ordinary telephone system, is a retronym for voice-grade telephone service employing analog signal transmission over copper loops. POTS was the standard service offering from telephone companies from 1 ...
(POTS).


Overview

The steps and principles involved in originating VoIP telephone calls are similar to traditional digital
telephony Telephony ( ) is the field of technology involving the development, application, and deployment of telecommunication services for the purpose of electronic transmission of voice, fax, or data, between distant parties. The history of telephony is i ...
and involve signaling, channel setup, digitization of the analog voice signals, and encoding. Instead of being transmitted over a
circuit-switched network Circuit switching is a method of implementing a telecommunications network in which two network nodes establish a dedicated communications channel ( circuit) through the network before the nodes may communicate. The circuit guarantees the full ...
, the digital information is packetized and transmission occurs as IP packets over a
packet-switched network In telecommunications, packet switching is a method of grouping data into '' packets'' that are transmitted over a digital network. Packets are made of a header and a payload. Data in the header is used by networking hardware to direct the pack ...
. They transport media streams using special media delivery protocols that encode audio and video with
audio codec An audio codec is a device or computer program capable of encoding or decoding a digital data stream (a codec) that encodes or decodes audio. In software, an audio codec is a computer program implementing an algorithm that compresses and decompres ...
s and
video codec A video codec is software or hardware that compresses and decompresses digital video. In the context of video compression, ''codec'' is a portmanteau of ''encoder'' and ''decoder'', while a device that only compresses is typically called an '' ...
s. Various codecs exist that optimize the media stream based on application requirements and network bandwidth; some implementations rely on
narrowband Narrowband signals are signals that occupy a narrow range of frequencies or that have a small fractional bandwidth. In the audio spectrum, narrowband sounds are sounds that occupy a narrow range of frequencies. In telephony, narrowband is usua ...
and compressed speech, while others support high-fidelity stereo codecs. The most widely used
speech coding Speech coding is an 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 da ...
standards in VoIP are based on the
linear predictive coding Linear predictive coding (LPC) is a method used mostly in audio signal processing and speech processing for representing the spectral envelope of a digital signal of speech in compressed form, using the information of a linear predictive model. ...
(LPC) and
modified discrete cosine transform The modified discrete cosine transform (MDCT) is a transform based on the type-IV discrete cosine transform (DCT-IV), with the additional property of being lapped transform, lapped: it is designed to be performed on consecutive blocks of a larger ...
(MDCT) compression methods. Popular codecs include the MDCT-based
AAC-LD The MPEG-4 Low Delay Audio Coder (a.k.a. AAC Low Delay, or AAC-LD) is audio compression standard designed to combine the advantages of perceptual audio coding with the low delay necessary for two-way communication. It is closely derived from the ...
(used in
FaceTime FaceTime is a Proprietary software, proprietary videotelephony product developed by Apple Inc. FaceTime is available on supported iOS mobile devices running iOS 4 and later and Mac computers that run and later. FaceTime supports any iOS devic ...
), the LPC/MDCT-based
Opus ''Opus'' (pl. ''opera'') is a Latin word meaning "work". Italian equivalents are ''opera'' (singular) and ''opere'' (pl.). Opus or OPUS may refer to: Arts and entertainment Music * Opus number, (abbr. Op.) specifying order of (usually) publicatio ...
(used in
WhatsApp WhatsApp (also called WhatsApp Messenger) is an internationally available freeware, cross-platform, centralized instant messaging (IM) and voice-over-IP (VoIP) service owned by American company Meta Platforms (formerly Facebook). It allows us ...
), the LPC-based
SILK Silk is a natural protein fiber, some forms of which can be woven into textiles. The protein fiber of silk is composed mainly of fibroin and is produced by certain insect larvae to form cocoons. The best-known silk is obtained from the coc ...
(used in
Skype Skype () is a proprietary telecommunications application operated by Skype Technologies, a division of Microsoft, best known for VoIP-based videotelephony, videoconferencing and voice calls. It also has instant messaging, file transfer, deb ...
), μ-law and
A-law An A-law algorithm is a standard companding algorithm, used in European 8-bit PCM digital communications systems to optimize, i.e. modify, the dynamic range of an analog signal for digitizing. It is one of two versions of the G.711 standard ...
versions of
G.711 G.711 is a narrowband audio codec originally designed for use in telephony that provides toll-quality audio at 64 kbit/s. G.711 passes audio signals in the range of 300–3400 Hz and samples them at the rate of 8,000 samples per second ...
,
G.722 G.722 is an ITU-T standard 7 kHz wideband audio codec operating at 48, 56 and 64 kbit/s. It was approved by ITU-T in November 1988. Technology of the codec is based on sub-band ADPCM (SB-ADPCM). The corresponding narrow-band codec based on ...
, and an
open source Open source is source code that is made freely available for possible modification and redistribution. Products include permission to use the source code, design documents, or content of the product. The open-source model is a decentralized sof ...
voice codec known as
iLBC Internet Low Bitrate Codec (iLBC) is a royalty-free narrowband speech audio coding format and an open-source reference implementation (codec), developed by Global IP Solutions (GIPS) formerly Global IP Sound (acquired by Google Inc in 2011). It w ...
, a codec that uses only 8 kbit/s each way called
G.729 G.729 is a royalty-free narrow-band vocoder-based audio data compression algorithm using a frame length of 10 milliseconds. It is officially described as ''Coding of speech at 8 kbit/s using code-excited linear prediction'' speech coding (CS-ACEL ...
. Early providers of voice-over-IP services used business models and offered technical solutions that mirrored the architecture of the legacy telephone network. Second-generation providers, such as
Skype Skype () is a proprietary telecommunications application operated by Skype Technologies, a division of Microsoft, best known for VoIP-based videotelephony, videoconferencing and voice calls. It also has instant messaging, file transfer, deb ...
, built closed networks for private user bases, offering the benefit of free calls and convenience while potentially charging for access to other communication networks, such as the PSTN. This limited the freedom of users to mix-and-match third-party hardware and software. Third-generation providers, such as
Google Talk Google Talk was an Instant messaging, instant messaging service that provided both text and voice communication. The instant messaging service was variously referred to colloquially as Gchat, Gtalk, or Gmessage among its users. Google Talk was ...
, adopted the concept of
federated VoIP Federated VoIP is a form of packetized voice telephony that uses voice over IP between autonomous domains in the public Internet without the deployment of central virtual exchange points or switching centers for traffic routing. Federated VoIP uses ...
. These solutions typically allow dynamic interconnection between users in any two domains of the Internet, when a user wishes to place a call. In addition to
VoIP phone A VoIP phone or IP phone uses voice over IP technologies for placing and transmitting telephone calls over an IP network, such as the Internet. This is in contrast to a standard phone which uses the traditional public switched telephone network ...
s, VoIP is also available on many personal computers and other Internet access devices. Calls and SMS text messages may be sent via
Wi-Fi Wi-Fi () is a family of wireless network protocols, based on the IEEE 802.11 family of standards, which are commonly used for local area networking of devices and Internet access, allowing nearby digital devices to exchange data by radio wave ...
or the carrier's
mobile data Mobile broadband is the marketing term for wireless Internet access via mobile networks. Access to the network can be made through a portable modem, wireless modem, or a tablet/ smartphone (possibly tethered) or other mobile device. The fi ...
network. VoIP provides a framework for consolidation of all modern communications technologies using a single
unified communications Unified communications (UC) is a business and marketing concept describing the integration of enterprise communication services such as instant messaging (chat), presence information, voice (including IP telephony), mobility features (including e ...
system.


Pronunciation

''VoIP'' is variously pronounced as an
initialism An acronym is a word or name formed from the initial components of a longer name or phrase. Acronyms are usually formed from the initial letters of words, as in ''NATO'' (''North Atlantic Treaty Organization''), but sometimes use syllables, as ...
, ''V-O-I-P'', or as an
acronym An acronym is a word or name formed from the initial components of a longer name or phrase. Acronyms are usually formed from the initial letters of words, as in ''NATO'' (''North Atlantic Treaty Organization''), but sometimes use syllables, as ...
, (). Full words, ''voice over Internet Protocol'', or ''voice over IP'', are sometimes used.


Protocols

Voice over IP has been implemented with
proprietary protocol In telecommunications, a proprietary protocol is a communications protocol owned by a single organization or individual. Intellectual property rights and enforcement Ownership by a single organization gives the owner the ability to place restricti ...
s and protocols based on
open standards An open standard is a standard that is openly accessible and usable by anyone. It is also a prerequisite to use open license, non-discrimination and extensibility. Typically, anybody can participate in the development. There is no single definition ...
in applications such as VoIP phones, mobile applications, and web-based communications. A variety of functions are needed to implement VoIP communication. Some protocols perform multiple functions, while others perform only a few and must be used in concert. These functions include: * ''Network'' and ''transport'' – Creating reliable transmission over unreliable protocols, which may involve acknowledging receipt of data and retransmitting data that wasn't received. * ''Session management'' – Creating and managing a session (sometimes glossed as simply a "call"), which is a connection between two or more peers that provides a context for further communication. * ''
Signaling In signal processing, a signal is a function that conveys information about a phenomenon. Any quantity that can vary over space or time can be used as a signal to share messages between observers. The ''IEEE Transactions on Signal Processing'' ...
'' – Performing registration (advertising one's presence and contact information) and discovery (locating someone and obtaining their contact information), dialing (including reporting call progress), negotiating capabilities, and call control (such as hold, mute, transfer/forwarding, dialing DTMF keys during a call .g._to_interact_with_an_automated_attendant_or_Interactive_voice_response.html" "title="automated_attendant.html" ;"title=".g. to interact with an automated attendant">.g. to interact with an automated attendant or Interactive voice response">IVR Interactive voice response (IVR) is a technology that allows telephone users to interact with a computer-operated telephone system through the use of voice and DTMF tones input with a keypad. In telecommunications, IVR allows customers to interac ...
], etc.). * ''Media description'' – Determining what type of media to send (audio, video, etc.), how to encode/decode it, and how to send/receive it (IP addresses, ports, etc.). * ''Media'' – Transferring the actual media in the call, such as audio, video, text messages, files, etc. * ''Quality of service'' – Providing out-of-band content or feedback about the media such as
synchronization Synchronization is the coordination of events to operate a system in unison. For example, the conductor of an orchestra keeps the orchestra synchronized or ''in time''. Systems that operate with all parts in synchrony are said to be synchronou ...
, statistics, etc. * ''Security'' – Implementing access control, verifying the identity of other participants (computers or people), and encrypting data to protect the privacy and integrity of the media contents and/or the control messages. VoIP protocols include: *
Session Initiation Protocol The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating communication sessions that include voice, video and messaging applications. SIP is used in Internet telephony, in private IP telepho ...
(SIP), connection management protocol developed by the IETF *
H.323 H.323 is a recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. The H.323 standard addresses call signaling and control, m ...
, one of the first VoIP call signaling and control protocols that found widespread implementation. 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. *
Media Gateway Control Protocol The Media Gateway Control Protocol (MGCP) is a signaling and call control communication protocol used in voice over IP (VoIP) telecommunication systems. It implements the media gateway control protocol architecture for controlling media gatewa ...
(MGCP), connection management for media gateways *
H.248 The Gateway Control Protocol (Megaco, H.248) is an implementation of the media gateway control protocol architecture for providing telecommunication services across a converged internetwork consisting of the traditional public switched telephone ...
, control protocol for media gateways across a converged internetwork consisting of the traditional PSTN and modern packet networks *
Real-time Transport Protocol The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks. RTP is used in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applicatio ...
(RTP), transport protocol for real-time audio and video data *
Real-time Transport Control Protocol The RTP Control Protocol (RTCP) is a sister protocol of the Real-time Transport Protocol (RTP). Its basic functionality and packet structure is defined in RFC 3550. RTCP provides out-of-band statistics and control information for an RTP session. ...
(RTCP), sister protocol for RTP providing stream statistics and status information *
Secure Real-time Transport Protocol The Secure Real-time Transport Protocol (SRTP) is a profile for Real-time Transport Protocol (RTP) intended to provide encryption, message authentication and integrity, and replay attack protection to the RTP data in both unicast and multicast ...
(SRTP), encrypted version of RTP *
Session Description Protocol The Session Description Protocol (SDP) is a format for describing multimedia communication sessions for the purposes of announcement and invitation. Its predominant use is in support of streaming media applications, such as voice over IP (VoIP) ...
(SDP), a syntax for session initiation and announcement for multi-media communications and
WebSocket WebSocket is a computer communications protocol, providing full-duplex communication channels over a single TCP connection. The WebSocket protocol was standardized by the IETF as in 2011. The current API specification allowing web applications ...
transports. *
Inter-Asterisk eXchange Inter-Asterisk eXchange (IAX) is a communications protocol native to the Asterisk private branch exchange (PBX) software, and is supported by a few other softswitches, PBX systems, and softphones. It is used for transporting VoIP telephony session ...
(IAX), protocol used between
Asterisk PBX Asterisk is a software implementation of a private branch exchange (PBX). In conjunction with suitable telephony hardware interfaces and network applications, Asterisk is used to establish and control telephone calls between telecommunication e ...
instances *
Extensible Messaging and Presence Protocol Extensible Messaging and Presence Protocol (XMPP, originally named Jabber) is an Open standard, open communication protocol designed for instant messaging (IM), presence information, and contact list maintenance. Based on XML (Extensible Markup ...
(XMPP), instant messaging, presence information, and contact list maintenance *
Jingle A jingle is a short song or tune used in advertising and for other commercial uses. Jingles are a form of sound branding. A jingle contains one or more hooks and meaning that explicitly promote the product or service being advertised, usually t ...
, for peer-to-peer session control in XMPP * Skype protocol, proprietary Internet telephony protocol suite based on peer-to-peer architecture


Adoption


Consumer market

Mass-market VoIP services use existing
broadband Internet access Internet access is the ability of individuals and organizations to connect to the Internet using computer terminals, computers, and other devices; and to access services such as email and the World Wide Web. Internet access is sold by Internet ...
, by which subscribers place and receive telephone calls in much the same manner as they would via the PSTN. Full-service VoIP phone companies provide inbound and outbound service with
direct inbound dialing Direct inward dialing (DID), also called direct dial-in (DDI) in Europe and Oceania, is a telecommunication service offered by telephone companies to subscribers who operate a private branch exchange (PBX) system. The feature provides service for ...
. Many offer unlimited domestic calling and sometimes international calls for a flat monthly subscription fee. 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
Ethernet Ethernet () is a family of wired computer networking technologies commonly used in local area networks (LAN), metropolitan area networks (MAN) and wide area networks (WAN). It was commercially introduced in 1980 and first standardized in 198 ...
or
Wi-Fi Wi-Fi () is a family of wireless network protocols, based on the IEEE 802.11 family of standards, which are commonly used for local area networking of devices and Internet access, allowing nearby digital devices to exchange data by radio wave ...
. These are typically designed in the style of traditional digital business telephones. * An
analog telephone adapter An analog telephone adapter (ATA) is a device for connecting traditional analog telephones, fax machines, and similar customer-premises devices to a digital telephone system or a voice over IP telephony network. An ATA is often built into a sma ...
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
cablemodem A cable modem is a type of network bridge that provides bi-directional data communication via radio frequency channels on a hybrid fibre-coaxial (HFC), radio frequency over glass (RFoG) and coaxial cable infrastructure. Cable modems are primari ...
s have this function built in. *
Softphone A softphone is a software program for making telephone calls over the Internet using a general purpose computer rather than dedicated hardware. The softphone can be installed on a piece of equipment such as a desktop, mobile device, or other compu ...
application software installed on a networked computer 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 increasingly common for telecommunications providers to use VoIP telephony over dedicated and public IP networks as a backhaul to connect switching centers and to interconnect with other telephony network providers; this is often referred to as ''IP backhaul''.
Smartphones A smartphone is a portable computer device that combines mobile telephone and computing functions into one unit. They are distinguished from feature phones by their stronger hardware capabilities and extensive mobile operating systems, which ...
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
Private branch exchange A business telephone system is a multiline telephone system typically used in business environments, encompassing systems ranging in technology from the key telephone system (KTS) to the private branch exchange (PBX). A business telephone syst ...
(PBX) lines installed internationally were VoIP. For example, in the United States, the
Social Security Administration The United States Social Security Administration (SSA) is an Independent agencies of the United States government, independent agency of the Federal government of the United States, U.S. federal government that administers Social Security (United ...
is converting its field offices of 63,000 workers from traditional phone installations to a VoIP infrastructure carried over its existing data network. 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
personal computer A personal computer (PC) is a multi-purpose microcomputer whose size, capabilities, and price make it feasible for individual use. Personal computers are intended to be operated directly by an end user, rather than by a computer expert or tec ...
s. 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 () is a family of wireless network protocols, based on the IEEE 802.11 family of standards, which are commonly used for local area networking of devices and Internet access, allowing nearby digital devices to exchange data by radio wave ...
network, so that it is no longer necessary to carry both a desktop phone and a cell phone. Maintenance becomes simpler as there are fewer devices to oversee. VoIP solutions aimed at businesses have evolved into
unified communications Unified communications (UC) is a business and marketing concept describing the integration of enterprise communication services such as instant messaging (chat), presence information, voice (including IP telephony), mobility features (including e ...
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 service providers are operating 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.
Skype Skype () is a proprietary telecommunications application operated by Skype Technologies, a division of Microsoft, best known for VoIP-based videotelephony, videoconferencing and voice calls. It also has instant messaging, file transfer, deb ...
, 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.


Delivery mechanisms

In general, the provision of VoIP telephony systems to organizational or individual users can be divided into two primary delivery methods: private or on-premises solutions, or externally hosted solutions delivered by third-party providers. On-premises delivery methods are more akin to the classic PBX deployment model for connecting an office to local PSTN networks. While many use cases still remain for private or on-premises VoIP systems, the wider market has been gradually shifting toward ''Cloud'' or ''Hosted'' VoIP solutions. Hosted systems are also generally better suited to smaller or personal use VoIP deployments, where a private system may not be viable for these scenarios.


Hosted VoIP systems

''Hosted'' or ''Cloud'' VoIP solutions involve a service provider or telecommunications carrier hosting the telephone system as a software solution within their own infrastructure. Typically this will be one or more datacentres, with geographic relevance to the end-user(s) of the system. This infrastructure is external to the user of the system and is deployed and maintained by the service provider. Endpoints, such as VoIP telephones or softphone applications (apps running on a computer or mobile device), will connect to the VoIP service remotely. These connections typically take place over public internet links, such as local fixed WAN breakout or mobile carrier service.


Private VoIP systems

In the case of a private VoIP system, the primary telephony system itself is located within the private infrastructure of the end-user organization. Usually, the system will be deployed on-premises at a site within the direct control of the organization. This can provide numerous benefits in terms of QoS control (see
below Below may refer to: *Earth *Ground (disambiguation) *Soil *Floor *Bottom (disambiguation) Bottom may refer to: Anatomy and sex * Bottom (BDSM), the partner in a BDSM who takes the passive, receiving, or obedient role, to that of the top or ...
), cost scalability, and ensuring privacy and security of communications traffic. However, the responsibility for ensuring that the VoIP system remains performant and resilient is predominantly vested in the end-user organization. This is not the case with a Hosted VoIP solution. Private VoIP systems can be physical hardware PBX appliances, converged with other infrastructure, or they can be deployed as software applications. Generally, the latter two options will be in the form of a separate virtualized appliance. However, in some scenarios, these systems are deployed on bare metal infrastructure or IoT devices. With some solutions, such as 3CX, companies can attempt to blend the benefits of hosted and private on-premises systems by implementing their own private solution but within an external environment. Examples can include
datacentre A data center (American English) or data centre (British English)See spelling differences. is a building, a dedicated space within a building, or a group of buildings used to house computer systems and associated components, such as telecommunic ...
collocation services, public cloud, or private cloud locations. For on-premises systems, local endpoints within the same location typically connect directly over the LAN. For remote and external endpoints, available connectivity options mirror those of Hosted or Cloud VoIP solutions. However, VoIP traffic to and from the on-premises systems can often also be sent over secure private links. Examples include personal VPN, site-to-site VPN, private networks such as MPLS and SD-WAN, or via private SBCs (Session Border Controllers). While exceptions and private peering options do exist, it is generally uncommon for those private connectivity methods to be provided by Hosted or Cloud VoIP providers.


Quality of service

Communication on the IP network is perceived as less reliable in contrast to the circuit-switched public telephone network because 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 service Quality of service (QoS) is the description or measurement of the overall performance of a service, such as a telephony or computer network, or a cloud computing service, particularly the performance seen by the users of the network. To quantitat ...
(QoS) guarantees. Voice, and all other data, travels in packets over IP networks with fixed maximum capacity. This system may be more prone to data loss in the presence of congestion than traditional
circuit switched Circuit switching is a method of implementing a telecommunications network in which two network nodes establish a dedicated communications channel ( circuit) through the network before the nodes may communicate. The circuit guarantees the full b ...
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. Therefore, VoIP implementations may face problems with latency, packet loss, and
jitter In electronics and telecommunications, jitter is the deviation from true periodicity of a presumably periodic signal, often in relation to a reference clock signal. In clock recovery applications it is called timing jitter. Jitter is a significa ...
. By default, network routers handle traffic on a first-come, first-served basis. 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 A geostationary orbit, also referred to as a geosynchronous equatorial orbit''Geostationary orbit'' and ''Geosynchronous (equatorial) orbit'' are used somewhat interchangeably in sources. (GEO), is a circular geosynchronous orbit in altitude ...
and back; delays of 400–600 ms are typical. Latency can be minimized by marking voice packets as being delay-sensitive with QoS methods such as
DiffServ Differentiated services or DiffServ is a computer networking architecture that specifies a mechanism for classifying and managing network traffic and providing quality of service (QoS) on modern IP networks. DiffServ can, for example, be used t ...
. Network routers on high volume traffic links may introduce latency that exceeds permissible thresholds for VoIP. Excessive load on a link can cause congestion and associated
queueing delay In telecommunication and computer engineering, the queuing delay or queueing delay is the time a job waits in a queue until it can be executed. It is a key component of network delay. In a switched network, queuing delay is the time between the c ...
s and
packet loss Packet loss occurs when one or more packets of data travelling across a computer network fail to reach their destination. Packet loss is either caused by errors in data transmission, typically across wireless networks, or network congestion.Kur ...
. 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 the link is congested by bulk traffic. VoIP endpoints usually have to wait for the completion of transmission of previous packets before new data may be sent. Although it is possible to preempt (abort) a less important packet in mid-transmission, 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
digital subscriber line Digital subscriber line (DSL; originally digital subscriber loop) is a family of technologies that are used to transmit digital data over telephone lines. In telecommunications marketing, the term DSL is widely understood to mean asymmetric di ...
(DSL), is to reduce the maximum transmission time by reducing the
maximum transmission unit In computer networking, the maximum transmission unit (MTU) is the size of the largest protocol data unit (PDU) that can be communicated in a single network layer transaction. The MTU relates to, but is not identical to the maximum frame size that ...
. But since every packet must contain protocol headers, this increases relative header overhead on every link traversed. The receiver must resequence IP packets that arrive out of order and recover gracefully when packets arrive too late or not at all.
Packet delay variation In computer networking, packet delay variation (PDV) is the difference in end-to-end one-way delay between selected packets in a flow with any lost packets being ignored.RFC 3393 The effect is sometimes referred to as packet jitter, although t ...
results from changes in
queuing delay In telecommunication and computer engineering, the queuing delay or queueing delay is the time a job waits in a queue until it can be executed. It is a key component of network delay. In a switched network, queuing delay is the time between the co ...
along a given network path due to competition from other users for the same transmission links. VoIP receivers accommodate this variation by storing incoming packets briefly in a
playout buffer In electronics and telecommunications, jitter is the deviation from true periodicity of a presumably periodic signal, often in relation to a reference clock signal. In clock recovery applications it is called timing jitter. Jitter is a significa ...
, deliberately increasing latency to improve the chance that each packet will be on hand when it is time for the
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. 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. Motivated by the
central limit theorem In probability theory, the central limit theorem (CLT) establishes that, in many situations, when independent random variables are summed up, their properly normalized sum tends toward a normal distribution even if the original variables themselv ...
, jitter can be modeled 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, 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 Transmission may refer to: Medicine, science and technology * Power transmission ** Electric power transmission ** Propulsion transmission, technology allowing controlled application of power *** Automatic transmission *** Manual transmission *** ...
medium (e.g. optical fiber) and the routers driving them do not have enough buffering for queuing delays to be significant. A number of protocols have been defined to support the reporting of
quality of service Quality of service (QoS) is the description or measurement of the overall performance of a service, such as a telephony or computer network, or a cloud computing service, particularly the performance seen by the users of the network. To quantitat ...
(QoS) and
quality of experience Quality of experience (QoE) is a measure of the delight or annoyance of a customer's experiences with a service (e.g., web browsing, phone call, TV broadcast).Qualinet White Paper on Definitions of Quality of Experience (2012). European Network on Q ...
(QoE) for VoIP calls. These include
RTP Control Protocol The RTP Control Protocol (RTCP) is a sister protocol of the Real-time Transport Protocol (RTP). Its basic functionality and packet structure is defined in RFC 3550. RTCP provides out-of-band statistics and control information for an RTP session. ...
(RTCP) extended reports, SIP RTCP summary reports, H.460.9 Annex B (for
H.323 H.323 is a recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. The H.323 standard addresses call signaling and control, m ...
),
H.248 The Gateway Control Protocol (Megaco, H.248) is an implementation of the media gateway control protocol architecture for providing telecommunication services across a converged internetwork consisting of the traditional public switched telephone ...
.30 and MGCP extensions. The RTCP extended report VoIP metrics block specified by is generated by an VoIP 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 (MOS) is a measure used in the domain of Quality of Experience and telecommunications engineering, representing overall quality of a stimulus or system. It is the arithmetic mean over all individual "values on a predefined scale t ...
s (MOS) and R factors and configuration information related to the jitter buffer. 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. 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.


DSL and ATM

DSL modems typically provide Ethernet connections to local equipment, but inside they may actually be
Asynchronous Transfer Mode Asynchronous Transfer Mode (ATM) is a telecommunications standard defined by American National Standards Institute (ANSI) and ITU-T (formerly CCITT) for digital transmission of multiple types of traffic. ATM was developed to meet the needs of ...
(ATM) modems. They use
ATM Adaptation Layer 5 ATM Adaptation Layer 5 (AAL5) is an ATM adaptation layer used to send variable-length packets up to 65,535 octets in size across an Asynchronous Transfer Mode (ATM) network. Unlike most network frames, which place control information in the he ...
(AAL5) to segment each Ethernet packet into a series of 53-byte ATM cells for transmission, reassembling them back into Ethernet frames at the receiving end. Using a separate
virtual circuit identifier Asynchronous Transfer Mode (ATM) is a telecommunications standard defined by American National Standards Institute (ANSI) and ITU-T (formerly CCITT) for digital transmission of multiple types of traffic. ATM was developed to meet the needs of ...
(VCI) for audio over IP has the potential to reduce latency on shared connections. 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 VCs. The latest generations of DSL,
VDSL Very high-speed digital subscriber line (VDSL) and very high-speed digital subscriber line 2 (VDSL2) are digital subscriber line (DSL) technologies providing data transmission faster than the earlier standards of asymmetric digital subscriber line ...
and
VDSL2 Very high-speed digital subscriber line (VDSL) and very high-speed digital subscriber line 2 (VDSL2) are digital subscriber line (DSL) technologies providing data transmission faster than the earlier standards of asymmetric digital subscriber lin ...
, 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. ATM has substantial header overhead: 5/53 = 9.4%, roughly twice the total header overhead of a 1500 byte Ethernet frame. This "ATM tax" is incurred by every DSL user whether or not they take advantage of multiple virtual circuits – and few can.


Layer 2

Several protocols are used in the
data link layer The data link layer, or layer 2, is the second layer of the seven-layer OSI model of computer networking. This layer is the protocol layer that transfers data between nodes on a network segment across the physical layer. The data link layer p ...
and
physical layer In the seven-layer OSI model of computer networking, the physical layer or layer 1 is the first and lowest layer; The layer most closely associated with the physical connection between devices. This layer may be implemented by a PHY chip. The ...
for quality-of-service mechanisms that help VoIP applications work well even in the presence of
network congestion Network congestion in data networking and queueing theory is the reduced quality of service that occurs when a network node or link is carrying more data than it can handle. Typical effects include queueing delay, packet loss or the blocking of ...
. Some examples include: *
IEEE 802.11e IEEE 802.11e-2005 or 802.11e is an approved amendment to the IEEE 802.11 standard that defines a set of quality of service (QoS) enhancements for wireless LAN applications through modifications to the media access control (MAC) layer.M. Benveni ...
is an approved amendment to the
IEEE 802.11 IEEE 802.11 is part of the IEEE 802 set of local area network (LAN) technical standards, and specifies the set of media access control (MAC) and physical layer (PHY) protocols for implementing wireless local area network (WLAN) computer commun ...
standard that defines a set of quality-of-service enhancements for wireless LAN applications through modifications to the Media Access Control (MAC) layer. The standard is considered of critical importance for delay-sensitive applications, such as voice over wireless IP. * IEEE 802.1p defines 8 different classes of service (including one dedicated to voice) for traffic on layer-2 wired
Ethernet Ethernet () is a family of wired computer networking technologies commonly used in local area networks (LAN), metropolitan area networks (MAN) and wide area networks (WAN). It was commercially introduced in 1980 and first standardized in 198 ...
. * The ITU-T G.hn standard, which provides a way to create a high-speed (up to 1 gigabit per second) Local area network (LAN) using existing home wiring (Power line communication, power lines, phone lines and Ethernet over coax, coaxial cables). G.hn provides QoS by means of Contention-Free Transmission Opportunities (CFTXOPs) which are allocated to flows (such as a VoIP call) that require QoS and which have negotiated a ''contract'' with the network controllers.


Performance metrics

The quality of voice transmission is characterized by several metrics that may be monitored by network elements and by the user agent hardware or software. Such metrics include network
packet loss Packet loss occurs when one or more packets of data travelling across a computer network fail to reach their destination. Packet loss is either caused by errors in data transmission, typically across wireless networks, or network congestion.Kur ...
, packet
jitter In electronics and telecommunications, jitter is the deviation from true periodicity of a presumably periodic signal, often in relation to a reference clock signal. In clock recovery applications it is called timing jitter. Jitter is a significa ...
, packet round-trip delay time, latency (delay), post-dial delay, and echo. The metrics are determined by VoIP performance testing and monitoring.


PSTN integration

A VoIP media gateway controller (aka Class-5 telephone switch, Class 5 Softswitch) works in cooperation with a media gateway (aka IP Business Gateway) and connects the digital media stream, so as to complete the path for voice and data. Gateways include interfaces for connecting to standard PSTN networks. Ethernet interfaces are also included in the modern systems which are specially designed to link calls that are passed via VoIP. E.164 is a global numbering standard for both the PSTN and public land mobile network (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 Technologies, Skype allows subscribers to choose ''Skype names'' (usernames) whereas SIP implementations can use Uniform Resource Identifier (URIs) similar to E-mail address, 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 E.164 number to URI mapping (ENUM) service in IMS and SIP. Echo can also be an issue for PSTN integration. Common causes of echo include Impedance matching, impedance mismatches in analog circuitry and an acoustic path from the receive to transmit signal at the receiving end.


Number portability

Local number portability (LNP) and mobile number portability (MNP) also impact VoIP business. 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. 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. A voice call originating in the VoIP environment also faces least-cost routing (LCR) challenges to reach its destination if the number is routed to a mobile phone number on a traditional mobile carrier. LCR 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 MNP 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 may be necessary to query the mobile network about which home network a mobile phone number belongs to. As the popularity of VoIP increases in the enterprise markets because of LCR options, VoIP needs to provide a certain level of reliability when handling calls.


Emergency calls

A telephone connected to a land line has a direct relationship between a telephone number and a physical location, which is maintained by the telephone company and available to emergency responders via the national emergency response service centers in form of emergency subscriber lists. When an emergency call is received by a center the location is automatically determined from its databases and displayed on the operator console. In IP telephony, no such direct link between location and communications end point exists. Even a provider having wired infrastructure, such as a DSL provider, may know only the approximate location of the device, based on the IP address allocated to the network router and the known service address. Some ISPs do not track the automatic assignment of IP addresses to customer equipment. IP communication provides for device mobility. For example, a residential broadband connection may be used as a link to a virtual private network of a corporate entity, in which case the IP address being used for customer communications may belong to the enterprise, not the residential ISP. Such off-premises extensions may appear as part of an upstream IP PBX. On mobile devices, e.g., a 3G handset or USB wireless broadband adapter, the IP address has no relationship with any physical location known to the telephony service provider, since a mobile user could be anywhere in a region with network coverage, even roaming via another cellular company. At the VoIP level, a phone or gateway may identify itself by its account credentials with a
Session Initiation Protocol The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating communication sessions that include voice, video and messaging applications. SIP is used in Internet telephony, in private IP telepho ...
(SIP) registrar. In such cases, the Internet telephony service provider (ITSP) knows only that a particular user's equipment is active. Service providers often provide emergency response services by agreement with the user who registers a physical location and agrees that, if an emergency number is called from the IP device, emergency services are provided to that address only. Such emergency services are provided by VoIP vendors in the United States by a system called Enhanced 911 (E911), based on the Wireless Communications and Public Safety Act. The VoIP E911 emergency-calling system associates a physical address with the calling party's telephone number. 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. "VoIP providers may not allow customers to opt-out of 911 service." The VoIP E911 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 accurate only if subscribers keep their emergency address information current.


Fax support

Sending
fax Fax (short for facsimile), sometimes called telecopying or telefax (the latter short for telefacsimile), is the telephonic transmission of scanned printed material (both text and images), normally to a telephone number connected to a printer o ...
es over VoIP networks is sometimes referred to as Fax over IP (FoIP). Transmission of fax documents was problematic in early VoIP implementations, as most voice digitization and compression codecs are optimized for the representation of the human voice and the proper timing of the modem signals cannot be guaranteed in a packet-based, connectionless network. A standards-based solution for reliably delivering fax-over-IP is the T.38 protocol. 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 may be a standard device connected to an
analog telephone adapter An analog telephone adapter (ATA) is a device for connecting traditional analog telephones, fax machines, and similar customer-premises devices to a digital telephone system or a voice over IP telephony network. An ATA is often built into a sma ...
(ATA), or it may be a software application or dedicated network device operating via an Ethernet interface. Originally, T.38 was designed to use UDP or TCP transmission methods across an IP network. Some newer high-end fax machines have built-in T.38 capabilities which are connected directly to a network switch or router. In T.38 each packet contains a portion of the data stream sent in the previous packet. Two successive packets have to be lost to actually lose data integrity.


Power requirements

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 electrical power. The susceptibility of phone service to power failures is a common problem even with traditional analog service where customers purchase 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. VoIP phones and VoIP telephone adapters connect to Network router, routers or cable modems which typically depend on the availability of mains electricity or locally generated power. Some VoIP service providers use customer premises equipment (e.g., cable modems) 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.


Security

Secure calls are possible using standardized protocols such as
Secure Real-time Transport Protocol The Secure Real-time Transport Protocol (SRTP) is a profile for Real-time Transport Protocol (RTP) intended to provide encryption, message authentication and integrity, and replay attack protection to the RTP data in both unicast and multicast ...
. Most of the facilities of creating a secure telephone connection over traditional phone lines, such as digitizing and digital transmission, are already in place with VoIP. It is necessary only to encrypt and authenticate the existing data stream. Automated software, such as a Business telephone system#Current trends, virtual PBX, may eliminate the need for personnel to greet and switch incoming calls. The security concerns for VoIP telephone systems are similar to those of other Internet-connected devices. This means that hackers with knowledge of VoIP vulnerabilities can perform denial-of-service attacks, harvest customer data, record conversations, and compromise voicemail messages. Compromised VoIP user account or session credentials may enable an attacker to incur substantial charges from third-party services, such as long-distance or international calling. The technical details of many VoIP protocols create challenges in routing VoIP traffic through firewall (networking), firewalls and network address translators, used to interconnect to transit networks or the Internet. Private session border controllers are often employed to enable VoIP calls to and from protected networks. Other methods to NAT traversal, traverse NAT devices involve assistive protocols such as STUN and Interactive Connectivity Establishment (ICE). Standards for securing VoIP are available in the
Secure Real-time Transport Protocol The Secure Real-time Transport Protocol (SRTP) is a profile for Real-time Transport Protocol (RTP) intended to provide encryption, message authentication and integrity, and replay attack protection to the RTP data in both unicast and multicast ...
(SRTP) and the ZRTP protocol for analog telephony adapters, as well as for some softphones. IPsec is available to secure Point-to-point (telecommunications), point-to-point VoIP at the transport level by using opportunistic encryption. Though many consumer VoIP solutions do not support encryption of the signaling path or the media, securing a VoIP phone is conceptually easier to implement using VoIP than on traditional telephone circuits. A result of the lack of widespread support for encryption is that it is relatively easy to eavesdrop on VoIP calls when access to the data network is possible. Free open-source solutions, such as Wireshark, facilitate capturing VoIP conversations. Government and military organizations use various security measures to protect VoIP traffic, such as voice over secure IP (VoSIP), secure voice over IP (SVoIP), and secure voice over secure IP (SVoSIP). The distinction lies in whether encryption is applied in the telephone endpoint or in the network. Secure voice over secure IP may be implemented by encrypting the media with protocols such as Secure Real-time Transport Protocol, SRTP and ZRTP. Secure voice over IP uses Type 1 encryption on a classified network, such as SIPRNet. Public Secure VoIP is also available with free GNU software and in many popular commercial VoIP programs via libraries, such as ZRTP. In June 2021, the National Security Agency, NSA (National Security Agency) released comprehensive documents describing the four attack planes of a communications system – the network, perimeter, session controllers and Communication endpoint, endpoints – and explaining security risks and mitigation techniques for each of them.


Caller ID

Voice over IP protocols and equipment provide caller ID support that is compatible with the PSTN. Many VoIP service providers also allow callers to configure custom caller ID information.


Hearing aid compatibility

Wireline telephones which are manufactured in, imported to, or intended to be used in the US with Voice over IP service, on or after February 28, 2020, are required to meet the hearing aid compatibility requirements set forth by the Federal Communications Commission.


Operational cost

VoIP has drastically reduced the cost of communication by sharing network infrastructure between data and voice. A single broadband connection has the ability to transmit multiple telephone calls.


Regulatory and legal issues

As the popularity of VoIP grows, governments are becoming more interested in regulating VoIP in a manner similar to PSTN services. Throughout the developing world, particularly in countries where regulation is weak or Regulatory capture, captured by the dominant operator, restrictions on the use of VoIP are often imposed, including in Panama where VoIP is taxed, Guyana where VoIP is prohibited. In Ethiopia, where the government is nationalizing telecommunication service, it is a criminal offense to offer services using VoIP. The country has installed firewalls to prevent international calls from being made using VoIP. These measures were taken after the popularity of VoIP reduced the income generated by the state-owned Telecommunications, telecommunication company.


Canada

In Canada, the Canadian Radio-television and Telecommunications Commission regulates telephone service, including VoIP telephony service. VoIP services operating in Canada are required to provide 9-1-1 emergency service.


European Union

In the European Union, the treatment of VoIP service providers is a decision for each national telecommunications 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). The relevant EU Directive is not clearly drafted concerning obligations that 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.


Arab states of the Gulf Cooperation Council, GCC


Oman

In Oman, it is illegal to provide or use unauthorized VoIP services, to the extent that web sites of unlicensed VoIP providers have been blocked. Violations may be punished with fines of 50,000 Omani Rial (about 130,317 US dollars), a two-year prison sentence or both. In 2009, police raided 121 Internet cafes throughout the country and arrested 212 people for using or providing VoIP services.


Saudi Arabia

In September 2017, Saudi Arabia lifted the ban on VoIPs, in an attempt to reduce operational costs and spur digital entrepreneurship.


United Arab Emirates

In the United Arab Emirates (UAE), it is illegal to provide or use unauthorized VoIP services. Web sites of unlicensed VoIP providers have been blocked. Some VoIP services such as
Skype Skype () is a proprietary telecommunications application operated by Skype Technologies, a division of Microsoft, best known for VoIP-based videotelephony, videoconferencing and voice calls. It also has instant messaging, file transfer, deb ...
were allowed. In January 2018, internet service providers in UAE blocked all VoIP apps, including Skype, but permitting only 2 government-approved VoIP apps (C’ME and BOTIM). In opposition, a petition on ''Change.org'' garnered over 5000 signatures, in response to which the website was blocked in UAE. On March 24, 2020, the United Arab Emirates loosened restriction on VoIP services earlier prohibited in the country, to ease communication during the COVID-19 pandemic. However, popular instant messaging applications like
WhatsApp WhatsApp (also called WhatsApp Messenger) is an internationally available freeware, cross-platform, centralized instant messaging (IM) and voice-over-IP (VoIP) service owned by American company Meta Platforms (formerly Facebook). It allows us ...
,
Skype Skype () is a proprietary telecommunications application operated by Skype Technologies, a division of Microsoft, best known for VoIP-based videotelephony, videoconferencing and voice calls. It also has instant messaging, file transfer, deb ...
, and
FaceTime FaceTime is a Proprietary software, proprietary videotelephony product developed by Apple Inc. FaceTime is available on supported iOS mobile devices running iOS 4 and later and Mac computers that run and later. FaceTime supports any iOS devic ...
remained blocked from being used for voice and video calls, constricting residents to use paid services from the country's state-owned telecom providers.


India

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 other computers but not to a normal phone number. Foreign-based VoIP server services are illegal to use in India. Internet telephony is permitted to the ISP with restrictions. The following services are permitted: # PC to PC; within or outside India # PC / a device / Adapter conforming to the standard of any international agencies like- ITU or IETF etc. in India to PSTN/PLMN abroad. # Any device / Adapter conforming to standards of International agencies like ITU, IETF etc. connected to ISP node with static IP address to similar device / Adapter; within or outside India. # Except whatever is described in , no other form of Internet Telephony is permitted. # In India no Separate Numbering Scheme is provided to the Internet Telephony. Presently the 10 digit Numbering allocation based on E.164 is permitted to the Fixed Telephony, GSM, CDMA wireless service. For Internet Telephony, the numbering scheme shall only conform to IP addressing Scheme of Internet Assigned Numbers Authority (IANA). Translation of E.164 number / private number to IP address allotted to any device and vice versa, by ISP to show compliance with IANA numbering scheme is not permitted. # The Internet Service Licensee is not permitted to have PSTN/PLMN connectivity. Voice communication to and from a telephone connected to PSTN/PLMN and following E.164 numbering is prohibited in India.


South Korea

In South 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 (USFK) 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 could continue to use their US-based VoIP subscription, but later arrivals are required to use a Korean-based VoIP provider, which by contract will offer pricing similar to the flat rates offered by US VoIP providers.


United States

In the United States, the Federal Communications Commission 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). Operators of ''Interconnected'' VoIP (fully connected to the PSTN) are mandated to provide Enhanced 911 service without special request, provide for customer location updates, clearly disclose any limitations on their E-911 functionality to their consumers, obtain affirmative acknowledgements of these disclosures from all consumers, and may not allow their customers to opt-out of 911 service. VoIP operators also receive the benefit of certain US telecommunications regulations, including an entitlement to interconnection and exchange of traffic with incumbent local exchange carriers via wholesale carriers. Providers of ''nomadic'' VoIP service—those who are unable to determine the location of their users—are exempt from state telecommunications regulation. Another legal issue that the United States Congress, 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 Federal 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.


History

The early developments of packet network designs by Paul Baran and other researchers were motivated by a desire for a higher degree of circuit redundancy and network availability in the face of infrastructure failures than was possible in the circuit-switched networks in telecommunications of the mid-twentieth century. Danny Cohen (engineer), Danny Cohen first demonstrated a form of packet telephony, packet voice in 1973 as part of a flight simulator application, which operated across the early ARPANET. On the early ARPANET, real-time voice communication was not possible with uncompressed pulse-code modulation (PCM) digital audio, digital speech packets, which had a bit rate of 64kbps, much greater than the 2.4kbps Internet bandwidth, bandwidth of early modems. The solution to this problem was
linear predictive coding Linear predictive coding (LPC) is a method used mostly in audio signal processing and speech processing for representing the spectral envelope of a digital signal of speech in compressed form, using the information of a linear predictive model. ...
(LPC), a
speech coding Speech coding is an 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 da ...
data compression algorithm that was first proposed by Fumitada Itakura of Nagoya University and Shuzo Saito of Nippon Telegraph and Telephone (NTT) in 1966. LPC was capable of speech compression down to 2.4kbps, leading to the first successful real-time conversation over ARPANET in 1974, between Culler-Harrison Incorporated in Goleta, California, and MIT Lincoln Laboratory in Lexington, Massachusetts. LPC has since been the most widely used speech coding method. Code-excited linear prediction (CELP), a type of LPC algorithm, was developed by Manfred R. Schroeder and Bishnu S. Atal in 1985.M. R. Schroeder and B. S. Atal, "Code-excited linear prediction (CELP): high-quality speech at very low bit rates," in ''Proceedings of the IEEE International Conference on Acoustics, Speech, and Signal Processing'' (ICASSP), vol. 10, pp. 937–940, 1985. LPC algorithms remain an audio coding standard in modern VoIP technology. In the following time span of about two decades, various forms of packet telephony were developed and industry interest groups formed to support the new technologies. Following the termination of the ARPANET project, and expansion of the
Internet The Internet (or internet) is the global system of interconnected computer networks that uses the Internet protocol suite (TCP/IP) to communicate between networks and devices. It is a '' network of networks'' that consists of private, pub ...
for commercial traffic, IP telephony was tested and deemed infeasible for commercial use until the introduction of VocalChat in the early 1990s and then in Feb 1995 the official release of Internet Phone (or iPhone for short) commercial software by VocalTec, based on th
Audio Transceiver
patent by Lior Haramaty and Alon Cohen, and followed by other VoIP infrastructure components such as telephony gateways and switching servers. Soon after it became an established area of interest in commercial labs of the major IT concerns. By the late 1990s, the first softswitches became available, and new protocols, such as
H.323 H.323 is a recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. The H.323 standard addresses call signaling and control, m ...
, MGCP and the
Session Initiation Protocol The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating communication sessions that include voice, video and messaging applications. SIP is used in Internet telephony, in private IP telepho ...
(SIP) gained widespread attention. In the early 2000s, the proliferation of high-bandwidth always-on Internet connections to residential dwellings and businesses, spawned an industry of Internet telephony service providers (ITSPs). The development of open-source telephony software, such as Asterisk (PBX), Asterisk PBX, fueled widespread interest and entrepreneurship in voice-over-IP services, applying new Internet technology paradigms, such as cloud services to telephony. In 1999, a discrete cosine transform (DCT) audio data compression algorithm called the
modified discrete cosine transform The modified discrete cosine transform (MDCT) is a transform based on the type-IV discrete cosine transform (DCT-IV), with the additional property of being lapped transform, lapped: it is designed to be performed on consecutive blocks of a larger ...
(MDCT) was adopted for the Siren (codec), Siren codec, used in the G.722.1 wideband audio coding standard. The same year, the MDCT was adapted into the LD-MDCT speech coding algorithm, used for the
AAC-LD The MPEG-4 Low Delay Audio Coder (a.k.a. AAC Low Delay, or AAC-LD) is audio compression standard designed to combine the advantages of perceptual audio coding with the low delay necessary for two-way communication. It is closely derived from the ...
format and intended for significantly improved audio quality in VoIP applications. MDCT has since been widely used in VoIP applications, such as the G.729.1 wideband codec introduced in 2006, Apple Inc., Apple's
FaceTime FaceTime is a Proprietary software, proprietary videotelephony product developed by Apple Inc. FaceTime is available on supported iOS mobile devices running iOS 4 and later and Mac computers that run and later. FaceTime supports any iOS devic ...
(using AAC-LD) introduced in 2010, the CELT codec introduced in 2011,Presentation of the CELT codec
by Timothy B. Terriberry (65 minutes of video, see als
presentation slides
in PDF) the
Opus ''Opus'' (pl. ''opera'') is a Latin word meaning "work". Italian equivalents are ''opera'' (singular) and ''opere'' (pl.). Opus or OPUS may refer to: Arts and entertainment Music * Opus number, (abbr. Op.) specifying order of (usually) publicatio ...
codec introduced in 2012, and
WhatsApp WhatsApp (also called WhatsApp Messenger) is an internationally available freeware, cross-platform, centralized instant messaging (IM) and voice-over-IP (VoIP) service owned by American company Meta Platforms (formerly Facebook). It allows us ...
's voice calling feature introduced in 2015.


Milestones

* 1966: Linear predictive coding (LPC) proposed by Fumitada Itakura of Nagoya University and Shuzo Saito of Nippon Telegraph and Telephone (NTT). * 1973: Packet telephony, Packet voice application by Danny Cohen (engineer), Danny Cohen. * 1974: The Institute of Electrical and Electronics Engineers (IEEE) publishes a paper entitled "A Protocol for Packet Network Interconnection". * 1974: Network Voice Protocol (NVP) tested over ARPANET in August 1974, carrying barely audible 16kpbs CVSD encoded voice. * 1974: The first successful real-time conversation over ARPANET achieved using 2.4kpbs LPC, between Culler-Harrison Incorporated in Goleta, California, and MIT Lincoln Laboratory in Lexington, Massachusetts. * 1977: Danny Cohen and Jon Postel of the USC Information Sciences Institute, and Vint Cerf of the Defense Advanced Research Projects Agency (DARPA), agree to separate IP from TCP, and create UDP for carrying real-time traffic. * 1981: IPv4 is described in RFC 791. * 1985: The National Science Foundation commissions the creation of National Science Foundation Network, NSFNET. * 1985: Code-excited linear prediction (CELP), a type of LPC algorithm, developed by Manfred R. Schroeder and Bishnu S. Atal. * 1986: Proposals from various standards organizations for VoATM, Voice over ATM, in addition to commercial packet voice products from companies such as StrataCom * 1991: Speak Freely, a voice-over-IP application, was released to the public domain. * 1992: The Frame Relay Forum conducts development of standards for Voice over Frame Relay. * 1992: InSoft Inc. announces and launches its desktop conferencing product Communique, which included VoIP and video. The company is credited with developing the first generation of commercial, US-based VoIP, Internet media streaming and real-time Internet telephony/collaborative software and standards that would provide the basis for the Real Time Streaming Protocol (RTSP) standard. * 1993 Release of VocalChat, a commercial packet network PC voice communication software from VocalTec. *1994: MTALK, a freeware LAN VoIP application for Linux * 1995: VocalTec releases ''Internet Phone'' commercial Internet phone software. ** Beginning in 1995, Intel, Microsoft and Radvision initiated standardization activities for VoIP communications system. * 1996: ** ITU-T begins development of standards for the transmission and signaling of voice communications over Internet Protocol networks with the
H.323 H.323 is a recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. The H.323 standard addresses call signaling and control, m ...
standard. ** US telecommunication companies petition the US Congress to ban Internet phone technology. **
G.729 G.729 is a royalty-free narrow-band vocoder-based audio data compression algorithm using a frame length of 10 milliseconds. It is officially described as ''Coding of speech at 8 kbit/s using code-excited linear prediction'' speech coding (CS-ACEL ...
speech codec introduced, using CELP (LPC) algorithm. * 1997: Level 3 Communications, Level 3 began development of its first softswitch, a term they coined in 1998. * 1999: ** The
Session Initiation Protocol The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating communication sessions that include voice, video and messaging applications. SIP is used in Internet telephony, in private IP telepho ...
(SIP) specification RFC 2543 is released. ** Mark Spencer (computer engineer), Mark Spencer of Digium develops the first Open source software, open source private branch exchange (PBX) software (Asterisk (PBX), Asterisk). ** A discrete cosine transform (DCT) variant called the
modified discrete cosine transform The modified discrete cosine transform (MDCT) is a transform based on the type-IV discrete cosine transform (DCT-IV), with the additional property of being lapped transform, lapped: it is designed to be performed on consecutive blocks of a larger ...
(MDCT) is adopted for the Siren (codec), Siren codec, used in the G.722.1 wideband audio coding standard. ** The MDCT is adapted into the LD-MDCT algorithm, used in the
AAC-LD The MPEG-4 Low Delay Audio Coder (a.k.a. AAC Low Delay, or AAC-LD) is audio compression standard designed to combine the advantages of perceptual audio coding with the low delay necessary for two-way communication. It is closely derived from the ...
standard. * 2001: INOC-DBA, first inter-provider Session Initiation Protocol, SIP network deployed; also first voice network to reach all seven continents. * 2003: First released in August 2003,
Skype Skype () is a proprietary telecommunications application operated by Skype Technologies, a division of Microsoft, best known for VoIP-based videotelephony, videoconferencing and voice calls. It also has instant messaging, file transfer, deb ...
was the creation of Niklas Zennström and Janus Friis, in cooperation with four Estonian developers. It quickly became a popular program that helped democratise VoIP. * 2004: Commercial VoIP service providers proliferate. * 2006: G.729.1 wideband codec introduced, using MDCT and CELP (LPC) algorithms. * 2007: VoIP device manufacturers and sellers boom in Asia, specifically in the Philippines where many families of overseas workers reside. * 2009:
SILK Silk is a natural protein fiber, some forms of which can be woven into textiles. The protein fiber of silk is composed mainly of fibroin and is produced by certain insect larvae to form cocoons. The best-known silk is obtained from the coc ...
codec introduced, using LPC algorithm,Audio-Mitschnitt
vom Treffen der IETF-Codec-Arbeitsgruppe auf der Konferenz IETF79 in Peking, China mit einer Darstellung der grundlegenden Funktionsprinzipien durch Koen Vos (MP3, ~70 MiB) and used for voice calling in
Skype Skype () is a proprietary telecommunications application operated by Skype Technologies, a division of Microsoft, best known for VoIP-based videotelephony, videoconferencing and voice calls. It also has instant messaging, file transfer, deb ...
. * 2010: Apple Inc., Apple introduces
FaceTime FaceTime is a Proprietary software, proprietary videotelephony product developed by Apple Inc. FaceTime is available on supported iOS mobile devices running iOS 4 and later and Mac computers that run and later. FaceTime supports any iOS devic ...
, which uses the LD-MDCT-based AAC-LD codec. * 2011: ** Rise of WebRTC technology which allows VoIP directly in browsers. ** CELT codec introduced, using MDCT algorithm. * 2012:
Opus ''Opus'' (pl. ''opera'') is a Latin word meaning "work". Italian equivalents are ''opera'' (singular) and ''opere'' (pl.). Opus or OPUS may refer to: Arts and entertainment Music * Opus number, (abbr. Op.) specifying order of (usually) publicatio ...
codec introduced, using MDCT and LPC algorithms.


See also

*Audio over IP *Communications Assistance For Law Enforcement Act *Comparison of audio network protocols *Comparison of VoIP software *Differentiated services *High bit rate audio video over Internet Protocol *Integrated services *Internet fax *IP Multimedia Subsystem *List of VoIP companies *Mobile VoIP *Network Voice Protocol *RTP payload formats *SIP trunking *UNIStim *VoIP VPN *VoiceXML *VoIP recording


Notes


References


External links

* * {{DEFAULTSORT:Voice over IP Voice over IP, Broadband Videotelephony Audio network protocols Office equipment