Voice over IP

Voice over Internet Protocol (VoIP), also known as IP telephony, is a set of technologies used mainly for making voice calls over Internet Protocol networks, such as the Internet. Instead of using traditional phone lines, VoIP sends voice as data packets through the Internet. This allows people to make and receive calls using services like Skype, Microsoft Teams, and Google Voice. Even regular phones can be used with VoIP by connecting them to the Internet using special devices called analog telephone adapters.

The terms Internet telephony, broadband telephony, and broadband phone service describe delivering voice and other communication services, such as fax, SMS, and voice messaging, over the Internet. This is different from the old system known as the public switched telephone network or plain old telephone service.

VoIP has grown to work with mobile phones too, through technologies like Voice over LTE (VoLTE) and Voice over NR (Vo5G). These let people make voice calls over mobile data networks using 4G and 5G connections, making VoIP an important part of today’s mobile world.

Overview

Voice over Internet Protocol (VoIP) is a technology that lets people make voice calls over the Internet. Instead of using traditional phone lines, VoIP sends voice as small pieces of data called packets through Internet networks. This works on many devices, including special VoIP phones, computers, and mobile devices connected via Wi-Fi or mobile data.

VoIP has evolved over time. Early services mimicked old phone systems, while newer ones, like Skype and Google Talk, allow users to connect freely across the Internet. Today, VoIP is also built into mobile networks like VoLTE for 4G and Vo5G for 5G, which let people make clear, high-quality calls using mobile phones while staying connected to regular phone numbers around the world.

Protocols

Voice over IP uses different systems to make voice calls work over the internet. Some of these systems are made by companies, while others follow rules that anyone can use. These systems help manage connections, share information about who is calling, and move the actual voice data.

There are many protocols, or sets of rules, used for VoIP. Some well-known ones include Session Initiation Protocol for managing connections, H.323 for controlling calls, and Real-time Transport Protocol for moving audio and video data quickly. Other protocols help with security, quality, and making sure the call works smoothly.

Adoption

VoIP services allow people to make phone calls using their regular internet connection. These services work with special VoIP phones, adapters for regular phones, or software programs on computers. Many companies offer plans with unlimited calling for a monthly fee, and calls between customers of the same service are often free.

Businesses often switch to VoIP to save money. It lets them use one network for both phone calls and data, which cuts costs. VoIP systems are flexible and can even let workers switch between office and mobile networks without dropping calls. These systems have grown to include many types of communication, all managed together.

Delivery mechanisms

VoIP systems can be set up in two main ways: private, on-premises solutions or hosted solutions provided by external companies. On-premises systems are like traditional office phone setups, placed directly within an organization's own network.

Many people now prefer hosted or cloud VoIP systems, especially for smaller uses or personal needs. These systems are managed by a service provider and hosted in their own data centers[/w/0], meaning the user doesn't have to worry about maintenance. Users connect to these services using VoIP phones or apps on computers and mobile devices over the internet. Private VoIP systems, on the other hand, are kept within the organization's own infrastructure, offering more control but also requiring the organization to handle maintenance and performance. These systems can be physical devices or software applications and often connect using local networks or secure private links like VPNs and private networks.

Quality of service

Communication over the internet can sometimes be less reliable than using regular phone lines because the internet does not always make sure that data arrives perfectly. Voice calls sent over the internet, called VoIP, travel in small pieces called packets. These packets might get lost or arrive out of order, which can make the call sound choppy.

To help with this, special methods can be used to make sure voice packets get through quickly, even when the internet is busy. For example, voice packets can be marked so they move ahead of other less important information. Some technologies help keep voice calls clear by managing how data is sent and received, making sure that voice sounds smooth even when the internet is crowded.

Performance metrics

The quality of voice transmission in VoIP can be measured using several important metrics. These include things like packet loss, jitter, latency (delay), post-dial delay, and echo. These metrics help ensure that voice calls sound clear and smooth when using VoIP services.

PSTN integration

A VoIP media gateway controller works with a media gateway to connect voice and data, linking calls made through VoIP to regular phone networks. Modern systems include Ethernet interfaces for these connections.

VoIP systems often use a global numbering standard to route calls between VoIP users and regular phones. They can also use other identification methods, like usernames or email-like addresses, to make and receive calls. Echo can sometimes be a problem, caused by differences in older phone circuitry or sound paths in the receiving device.

Number portability

Local and mobile number portability let people keep their phone numbers when switching providers. VoIP providers must follow rules to make sure calls are correctly routed, even when numbers are moved between carriers. This can be complex, especially when calls are made to mobile phones on traditional networks.

Emergency calls

Traditional landlines have a direct link between a phone number and a physical location, helping emergency services find callers. With VoIP, this link isn't automatic because devices can move and use different networks. In the United States, VoIP providers use a system called Enhanced 911 (E911) to connect phone numbers with physical addresses for emergency calls. This system works only if users keep their address information up to date.

Fax support

Sending faxes over VoIP networks is called Fax over IP (FoIP). Early VoIP systems had trouble sending fax documents because the ways they changed voice into digital signals were not good for fax machines.

The T.38 protocol helps solve this problem. It makes sure fax messages can travel over IP networks, even though they work differently from old analog lines. Fax machines can be regular ones connected to an analog telephone adapter (ATA) or special software or devices connected to a network. Some newer high-end fax machines can use T.38 directly and are connected to a network switch or router.

Power requirements

Traditional home phones are often connected to special phone lines that give them power directly, so they work even if the house power is off. However, these phones can still stop working if there is a problem with the phone lines.

VoIP phones and adapters connect to devices like routers or cable modems, which need regular house power to work. Some VoIP services have special equipment with batteries that can keep the phone working for a few hours if the power goes out. If the internet device can't be used, some services can send calls to a cell phone instead.

Security

VoIP calls can be kept secure using special protocols like Secure Real-time Transport Protocol. Like regular phones, VoIP systems already have many tools for digital communication; they just need extra encryption and verification to keep data safe.

VoIP systems face similar security risks as other internet devices. Hackers might try to disrupt service, steal information, or listen in on conversations. Special software and tools can help protect VoIP calls, especially for governments and military groups, by using strong encryption methods.

Caller ID

Voice over IP systems support caller ID just like regular phone networks do. This means when someone calls you, their number or name can show up on your phone or device. Many VoIP services let you choose what information appears as your caller ID when you make a call.

Hearing aid compatibility

Wireline telephones made or used in the United States with Voice over IP service since February 28, 2020, must meet special requirements for hearing aid compatibility. These rules are set by the Federal Communications Commission to ensure that people who use hearing aids can still use their devices with modern telephones.

Operational cost

VoIP has made communication much cheaper by using the same network for both data and voice. With just one broadband connection, you can make and receive many phone calls at once. This helps save money compared to traditional phone services.

Regulatory and legal issues

As VoIP becomes more popular, governments are taking more interest in regulating it similarly to traditional phone services.

In some countries, like Panama and Ethiopia, there are strict rules or even bans on VoIP. In Canada, VoIP services must provide 9-1-1 emergency service. In the European Union, each country decides how to regulate VoIP, usually separating services that use managed networks from those using the general Internet.

In Saudi Arabia, a ban on VoIP was lifted in 2017 to help businesses. In the United Arab Emirates, using unauthorized VoIP services is against the law, though some services like Skype were allowed until 2018 when many were blocked. Restrictions were relaxed in 2020 due to the COVID-19 pandemic, but popular apps like WhatsApp and FaceTime remained blocked for voice and video calls.

In the United States, VoIP providers must follow rules similar to traditional phone companies, including supporting local number portability, making services accessible to people with disabilities, and helping law enforcement with surveillance when needed.

History

The history of Voice over Internet Protocol (VoIP) began with early researchers exploring ways to make voice calls over networks. In the 1970s, scientists worked on sending voice as data packets, which was difficult at the time because of limited internet speeds. They solved this by creating special methods to compress voice data, making it possible to send voice over the early internet.

Over the years, many new technologies and programs were developed to improve VoIP. By the 1990s, commercial VoIP software began appearing, and in the 2000s, popular programs like Skype made VoIP easy for everyone to use. Today, VoIP is a common way to make voice calls over the internet, using many of the ideas and technologies invented in these early years.

Milestones

• 1966: Linear predictive coding (LPC) proposed by Fumitada Itakura of Nagoya University and Shuzo Saito of Nippon Telegraph and Telephone (NTT).
• 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".
• 1974: Network Voice Protocol (NVP) tested over ARPANET in August 1974, carrying barely intelligible 16 kpbs CVSD encoded voice.
• 1974: The first successful real-time conversation over ARPANET was achieved using 2.4 kpbs 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 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^[specify] for 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 includes 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.^{[citation needed]}
• 1993 Release of VocalChat, a commercial packet network PC voice communication software from VocalTec.^{[citation needed]}
• 1994: MTALK, a freeware LAN VoIP application for Linux
• 1995:
  • VocalTec releases Internet Phone commercial Internet phone software.
  • 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 standard.
  • US telecommunications companies petition the US Congress to ban Internet phone technology.
  • G.729 speech codec introduced, using CELP (LPC) algorithm.
• 1997: Level 3 began development of its first softswitch, a term they coined in 1998.
• 1999:
  • The Session Initiation Protocol (SIP) specification RFC 2543 is released.
  • Mark Spencer of Digium develops Asterisk, the first open source private branch exchange (PBX) software.
  • 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.
  • The MDCT is adapted into the LD-MDCT algorithm, used in the AAC-LD standard.
• 2001: INOC-DBA, the first inter-provider SIP network is deployed; this is also the first voice network to reach all seven continents.
• 2003: Skype released in August 2003. This was the creation of Niklas Zennström and Janus Friis, in cooperation with four Estonian developers. It quickly became a popular program that helped democratize VoIP.
• 2004: Early commercial VoIP service providers proliferate.^{[citation needed]}
• 2005: PhoneGnome VoIP service is launched by TelEvolution, Inc. of California.
• 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 codec introduced, using LPC algorithm, and used for voice calling in Skype.
• 2010: Apple introduces FaceTime, which uses the LD-MDCT-based AAC-LD codec.
• 2011:
  • Rise of WebRTC technology, which supports VoIP directly in browsers.
  • CELT codec introduced, using MDCT algorithm.
• 2012: Opus codec introduced, using MDCT and LPC algorithms.