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VoIP

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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 communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. The terms Internet telephony, broadband telephony, and broadband phone service specifically refer to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN), also known as plain old telephone service (POTS).

The steps and principles involved in originating VoIP telephone calls are similar to traditional digital telephony and involve signaling, channel setup, digitization of the analog voice signals, and encoding. Instead of being transmitted over a circuit-switched network, the digital information is packetized and transmission occurs as IP packets over a packet-switched network. They transport media streams using special media delivery protocols that encode audio and video with audio codecs and video codecs. Various codecs exist that optimize the media stream based on application requirements and network bandwidth; some implementations rely on narrowband and compressed speech, while others support high-fidelity stereo codecs.

The most widely speech coding standards in VoIP are based on the linear predictive coding (LPC) and modified discrete cosine transform (MDCT) compression methods. Popular codecs include the MDCT-based AAC-LD (used in FaceTime), the LPC/MDCT-based Opus (used in WhatsApp), the LPC-based SILK (used in Skype), μ-law and A-law versions of G.711, G.722, and an open source voice codec known as iLBC, a codec that uses only 8 kbit/s each way called G.729.

Early providers of voice-over-IP services offered business models and technical solutions that mirrored the architecture of the legacy telephone network. Second-generation providers, such as Skype, 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, adopted the concept of federated VoIP—which is a departure from the architecture of the legacy networks.[1] These solutions typically allow dynamic interconnection between users on any two domains on the Internet when a user wishes to place a call.

In addition to VoIP phones, VoIP is also available on many personal computers and other Internet access devices. Calls and SMS text messages may be sent over mobile data or Wi-Fi.[2] VoIP allows modern communications technologies (including telephones, smartphones, voice and video conferencing, email, and presence detection) to be consolidated using a single unified communications system.

Pronunciation[]

VoIP is variously pronounced as an initialism, V-O-I-P, or as an acronym /ˈvɔɪp/ (voyp), as in voice.[3] Full words, voice over Internet Protocol, or voice over IP, are sometimes used.

Protocols[]

Voice over IP has been implemented in various ways using both proprietary protocols and protocols based on open standards. These protocols can be used by a VoIP phone, special-purpose software, a mobile application or integrated into a web page. VoIP protocols include:

  • Session Initiation Protocol (SIP), connection management protocol developed by the IETF
  • H.323, one of the first VoIP call signaling and control protocols that found widespread implementation.
  • Media Gateway Control Protocol (MGCP), connection management for media gateways
  • H.248, control protocol for media gateways across a converged internetwork consisting of the traditional public switched telephone network (PSTN) and modern packet networks
  • Real-time Transport Protocol (RTP), transport protocol for real-time audio and video data
  • Real-time Transport Control Protocol (RTCP), sister protocol for RTP providing stream statistics and status information
  • Secure Real-time Transport Protocol (SRTP), encrypted version of RTP
  • Session Description Protocol (SDP), file format used principally by SIP to describe VoIP connections
  • Inter-Asterisk eXchange (IAX), protocol used between VoIP servers
  • Extensible Messaging and Presence Protocol (XMPP), instant messaging, presence information, and contact list maintenance
  • Jingle, adds peer-to-peer session control to XMPP
  • Skype protocol, proprietary Internet telephony protocol suite based on peer-to-peer architecture

Adoption[]

Consumer market[]

Voip-typical

Example of residential network including VoIP

Mass-market VoIP services use existing broadband Internet access, 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 and sometimes international calls for a flat monthly subscription fee.

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 or Wi-Fi. These are typically designed in the style of traditional digital business telephones.
  • An analog telephone adapter 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.
  • Softphone application software installed on a networked computer that is equipped with a microphone and speaker, or headset.

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.[4][5]

Smartphones may have SIP clients built into the firmware or available as an application download.[6][7]

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 (PBX) lines installed internationally were VoIP.[8] For example, in the United States, the Social Security Administration is converting its field offices of 63,000 workers from traditional phone installations to a VoIP infrastructure carried over its existing data network.[9][10]

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 computers. Rather than closed architectures, these devices rely on standard interfaces.[11] 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 cell phone. Maintenance becomes simpler as there are fewer devices to oversee.[11]

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 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.[12]

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.[13]

Performance metrics[]

The quality of voice transmission is characterized by several metrics that may be monitored by network elements, by the user agent hardware or software. Such metrics include network packet loss, packet jitter, packet latency (delay), post-dial delay, and echo. The metrics are determined by VoIP performance testing and monitoring.[14][15][16][17][18][19]

Fax support[]

Sending faxes 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, connection-less 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 (ATA), or it may be a software application or dedicated network device operating via an Ethernet interface.[20] Originally, T.38 was designed to use UDP or TCP transmission methods across an IP network. UDP provides near real-time characteristics due to the "no recovery rule" when a UDP packet is lost or an error occurs during transmission.[21]

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.

IP Phones and VoIP telephone adapters connect to routers or cable modems which typically depend on the availability of mains electricity or locally generated power.[22] Some VoIP service providers use customer premises 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.

Security[]

The security concerns of 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 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 traverse NAT devices involve assistive protocols such as STUN and Interactive Connectivity Establishment (ICE).

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 than on traditional telephone circuits. A result of the lack of encryption is that it is relatively easy to eavesdrop on VoIP calls when access to the data network is possible.[23] Free open-source solutions, such as Wireshark, facilitate capturing VoIP conversations.

Standards for securing VoIP are available in the Secure Real-time Transport Protocol (SRTP) and the ZRTP protocol for analog telephony adapters, as well as for some softphones. IPsec is available to secure point-to-point VoIP at the transport level by using opportunistic encryption.

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).[24] The distinction lies in whether encryption is applied in the telephone endpoint or in the network.[25] Secure voice over secure IP may be implemented by encrypting the media with protocols such as SRTP and ZRTP. Secure voice over IP uses Type 1 encryption on a classified network, such as SIPRNet.[26][27][28][29] Public Secure VoIP is also available with free GNU software and in many popular commercial VoIP programs via libraries, such as ZRTP.[30]

Caller ID[]

Voice over IP protocols and equipment provide caller ID support that is compatible with the facility provided in the public switched telephone network (PSTN). Many VoIP service providers also allow callers to configure arbitrary caller ID information.[31]

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.[32]

Operational cost[]

VoIP has drastically reduced the cost of communication by sharing network infrastructure between data and voice.[33][34] A single broad-band connection has the ability to transmit more than one telephone call. Secure calls using standardized protocols, such as Secure Real-time Transport Protocol, as 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 virtual PBX, may eliminate the need of personnel to greet and switch incoming 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.[35]

Throughout the developing world, particularly in countries where regulation is weak or 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.[36] In Ethiopia, where the government is nationalising telecommunication service, it is a criminal offence 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.

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.[37]

Arab states of the 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.[38]

Saudi Arabia[]

In September 2017, Saudi Arabia lifted the ban on VoIPs, in an attempt to reduce operational costs and spur digital entrepreneurship.[39][40]

United Arab Emirates[]

In the United Arab Emirates (UAE), it is illegal to provide or use unauthorized VoIP services, to the extent that web sites of unlicensed VoIP providers have been blocked. However, some VoIPs such as Skype were allowed.[41] 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) for a fixed rate of Dh52.50 a month for use on mobile devices, and Dh105 a month to use over a computer connected."[42][43] In opposition, a petition on Change.org garnered over 5000 signatures, in response to which the website was blocked in UAE.[44]

India[]

In India, it is legal to use VoIP, but it is illegal to have VoIP gateways inside India.[45] 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 is not permitted by law to be inside India. Foreign based VoIP server services are illegal to use in India.[45]

In the interest of the Access Service Providers and International Long Distance Operators the Internet telephony was permitted to the ISP with restrictions. Internet Telephony is considered to be different service in its scope, nature and kind from real time voice as offered by other Access Service Providers and Long Distance Carriers. Hence the following type of Internet Telephony are permitted in India:[46]

(a) PC to PC; within or outside India
(b) PC / a device / Adapter conforming to standard of any international agencies like- ITU or IETF etc. in India to PSTN/PLMN abroad.
(c) 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.
(d) Except whatever is described in condition (ii) above, no other form of Internet Telephony is permitted.
(e) 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.
(f) 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 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.[47]

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.[48] 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,[49] and 'may not allow their customers to “opt-out” of 911 service.'[50] 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.[51]

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 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.[52]

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 first demonstrated a form of packet voice in 1973 as part of a flight simulator application, which operated across the early ARPANET.[53][54]

On the early ARPANET, real-time voice communication was not possible with uncompressed pulse-code modulation (PCM) digital speech packets, which had a bit rate of 64 kbps, much greater than the 2.4 kbps bandwidth of early modems. The solution to this problem was linear predictive coding (LPC), a speech coding 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 2.4 kbps, 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.[55] LPC has since been the most widely used speech coding method.[56] Code-excited linear prediction (CELP), a type of LPC algorithm, was developed by Manfred R. Schroeder and Bishnu S. Atal in 1985.[57] LPC algorithms remain an audio coding standard in modern VoIP technology.[55]

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 for commercial traffic, IP telephony became an established area of interest in commercial labs of the major IT concerns, such Microsoft and Intel, and open-source software, such as VocalTec, became available by the mid-1990s. By the late 1990s, the first softswitches became available, and new protocols, such as H.323, MGCP and the Session Initiation Protocol (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, 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 (MDCT) was adopted for the Siren codec, used in the G.722.1 wideband audio coding standard.[58][59] The same year, the MDCT was adapted into the LD-MDCT speech coding algorithm, used for the AAC-LD format and intended for significantly improved audio quality in VoIP applications.[60] MDCT has since been widely used in VoIP applications, such as the G.729.1 wideband codec introduced in 2006,[61] Apple's Facetime (using AAC-LD) introduced in 2010,[62] the CELT codec introduced in 2011,[63] the Opus codec introduced in 2012,[64] and WhatsApp's voice calling feature introduced in 2015.[65]

Milestones[]

  • 1966: Linear predictive coding (LPC) proposed by Fumitada Itakura of Nagoya University and Shuzo Saito of Nippon Telegraph and Telephone (NTT).[55]
  • 1973: Packet voice application by Danny Cohen.
  • 1974: The Institute of Electrical and Electronics Engineers (IEEE) publishes a paper entitled "A Protocol for Packet Network Interconnection".[66]
  • 1974: Network Voice Protocol (NVP) tested over ARPANET in August 1974, carrying barely audible 16 kpbs CVSD encoded voice.[55]
  • 1974: The first successful real-time conversation over ARPANET achieved using 2.4 kpbs LPC, between Culler-Harrison Incorporated in Goleta, California, and MIT Lincoln Laboratory in Lexington, Massachusetts.[55]
  • 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 NSFNET.[67]
  • 1985: Code-excited linear prediction (CELP), a type of LPC algorithm, developed by Manfred R. Schroeder and Bishnu S. Atal.[57]
  • 1991: First Voice-over-IP application, Speak Freely, is released into the public domain. It was originally written by John Walker and further developed by Brian C. Wiles.[68]
  • 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.[69] 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.[70][71]
  • 1994: MTALK, a freeware VoIP application for Linux[72]
  • 1995: VocalTec releases Internet Phone commercial Internet phone software.[73][74]
    • Beginning in 1995, Intel, Microsoft and Radvision initiated standardization activities for VoIP communications system.[75]
  • 1996:
    • ITU-T begins development of standards for the transmission and signaling of voice communications over Internet Protocol networks with the H.323 standard.[76]
    • US telecommunication companies petition the US Congress to ban Internet phone technology.[77]
    • G.729 speech codec introduced, using CELP (LPC) algorithm.[78]
  • 1997: Level 3 began development of its first softswitch, a term they coined in 1998.[79]
  • 1999:
    • The Session Initiation Protocol (SIP) specification RFC 2543 is released.[80]
    • Mark Spencer of Digium develops the first open source private branch exchange (PBX) software (Asterisk).[81]
    • A discrete cosine transform (DCT) variant called the modified discrete cosine transform (MDCT) is adopted for the Siren codec, used in the G.722.1 wideband audio coding standard.[58][59]
    • The MDCT is adapted into the LD-MDCT algorithm, used in the AAC-LD standard.[60]
  • 2004: Commercial VoIP service providers proliferate.
  • 2006: G.729.1 wideband codec introduced, using MDCT and CELP (LPC) algorithms.[61]
  • 2007: VoIP device manufacturers and sellers boom in Asia, specifically in the Philippines where many families of overseas workers reside.[82]
  • 2009: SILK codec introduced, using LPC algorithm,[83] and used for voice calling in Skype.[84]
  • 2010: Apple introduces FaceTime, which uses the LD-MDCT-based AAC-LD codec.[62]
  • 2011:
    • Rise of WebRTC technology which allows VoIP directly in browsers.
    • CELT codec introduced, using MDCT algorithm.[63]
  • 2012: Opus codec introduced, using MDCT and LPC algorithms.[64]
  • 2015: WhatsApp introduces voice calling feature, using the MDCT/LPC-based Opus codec.[65]
  • 2018: WhatsApp reaches over 1.5 billion users worldwide.[85]

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 audio video profile
  • SIP Trunking
  • UNIStim
  • Voice VPN
  • VoiceXML
  • VoIP recording

References[]

  1. ↑ "XMPP Federation". Google Talkabout. 2006. Retrieved 2012-05-11. {{cite web}}:
  2. ↑ Booth, C (2010). "Chapter 2: IP Phones, Software VoIP, and Integrated and Mobile VoIP". Library Technology Reports 46 (5): 11–19. 
  3. ↑ "VoIP". VoIP. http://dictionary.cambridge.org/pronunciation/british/voip. 
  4. ↑ "WIRELESS: Carriers look to IP for backhaul". www.eetimes.com. EE Times. Archived from the original on August 9, 2011. Retrieved 8 April 2015. {{cite web}}: CS1 maint: unfit URL (link)
  5. ↑ "Mobile's IP challenge". www.totaltele.com. Total Telecom Online. Archived from the original on February 17, 2006. Retrieved 8 April 2015. {{cite web}}: CS1 maint: unfit URL (link)
  6. ↑ "Android SIP Client". Retrieved 2018-01-30. {{cite web}}:
  7. ↑ "Learn to Make Free or Inexpensive Calls Using SIP on Android". Retrieved 2018-01-30. {{cite web}}:
  8. ↑ Michael Dosch and Steve Church. "VoIP in the Broadcast Studio". Axia Audio. Archived from the original on October 7, 2011. Retrieved June 21, 2011. {{cite web}}:
  9. ↑ Jackson, William (May 27, 2009). "SSA goes big on VOIP". Government Computer News. Retrieved 2009-05-28. {{cite web}}:
  10. ↑ "Social Security to Build "World's Largest VOIP"". Government Technology. Archived from the original on 2 June 2009. Retrieved 2009-05-29. {{cite web}}:
  11. ↑ 11.0 11.1 Korzeniowski, Peter (January 8, 2009). "Three Technologies You Need In 2009". Forbes. https://www.forbes.com/2009/01/08/small-business-voip-ent-tech-cx_bm_0108bmightytech09.html/. Retrieved 2009-03-02. 
  12. ↑ Callahan, Renee (December 9, 2008). "Businesses Move To Voice-Over-IP". Forbes. https://www.forbes.com/2008/12/09/skype-vonage-ringcentral_leadership_clayton_in_rc_1209claytonchristensen_inl.html. Retrieved 2009-03-03. 
  13. ↑ "Skype For Business". skype.com. Retrieved 2009-03-16. {{cite web}}:
  14. ↑ CableLabs, PacketCable Residential SIP Telephony Feature Definition, Technical Report, PKT-TR-RST-V03-071106 (2007)
  15. ↑ "VoIP performance measurement using QoS parameters" (PDF). A.H.Muhamad Amin. 2016-08-14. {{cite web}}:
  16. ↑ "Methodology for SIP Infrastructure Performance Testing" (PDF). Miroslav Voznak, Jan Rozhon. 2016-08-14. {{cite web}}:
  17. ↑ "Voice over IP (VoIP) Performance Evaluation on VMware vSphere® 5" (PDF). VMware. 2016-08-14. {{cite web}}:
  18. ↑ "Performance and Stress Testing of SIP Servers, Clients and IP Networks". StarTrinity. 2016-08-13. {{cite web}}:
  19. ↑ "Testing Voice over IP (VolP) Networks" (PDF). IXIA. 2016-08-14. {{cite web}}:
  20. ↑ Soft-Switch.org, Faxing over IP networks
  21. ↑ "UMass Discussion on UDP transmission Characteristics". {{cite web}}:
  22. ↑ "ICT Regulation Tool Kit – 4.4 VOIP – Regulatory Issues – Universal Service". Retrieved September 21, 2017. {{cite web}}:
  23. ↑ "Examining Two Well-Known Attacks on VoIP". CircleID. Retrieved 2006-04-05. {{cite web}}:
  24. ↑ Disa.mil, Internet Protocol Telephony & Voice over Internet Protocol Security Technical Implementation Guide
  25. ↑ Secure Voice over IP (SVoIP) vs. Voice over Secure IP (VoSIP) Installations General Dynamics C4 Systems
  26. ↑ Dunte, Markus; Ruland, Christoph (June 2007). "Secure Voice-over-IP". International Journal of Computer Science and Network Security 7 (6): 63–68. http://paper.ijcsns.org/07_book/200706/20070610.pdf. 
  27. ↑ Sans.org, SANS Institute InfoSec Reading Room
  28. ↑ White, C.M.; Teague, K.A.; Daniel, E.J. (7–10 Nov 2004). Browse Conference Publications > Signals, Systems and Computer ... Help Working with Abstracts Packet loss concealment in a secure voice over IP environment. 1. pp. 415–419. doi:10.1109/ACSSC.2004.1399165. ISBN 978-0-7803-8622-8. http://www.clsp.jhu.edu/~cwhite/papers/asilo_04_LossConceal_final.pdf. 
  29. ↑ "Cellcrypt secure VOIP heading to BlackBerry". Networkworld.com. {{cite web}}:
  30. ↑ "Secure VOIP calling, free software, and the right to privacy". Free Software Magazine. {{cite web}}:
  31. ↑ VOIPSA.org, Blog: "Hello Mom, I'm a Fake!" (Telespoof and Fakecaller).
  32. ↑ "Hearing Aid Compatibility for Wireline and Wireless Telephones". Federal Communications Commission. 2014-10-30. Retrieved 2019-07-09. {{cite web}}:
  33. ↑ FCC.gov, What are some advantages of VoIP?
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  35. ↑ "Global VOIP Policy Status Matrix". Global IP Alliance. Retrieved 2006-11-23. {{cite web}}:
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