Voice over IP
Voice over Internet Protocol, also known as IP telephony, is a set of technologies used primarily for voice communication sessions over Internet Protocol networks, such as the Internet. VoIP enables voice calls to be transmitted as data packets, facilitating various methods of voice communication, including traditional applications like Skype, Microsoft Teams, Google Voice, and VoIP phones. Regular telephones can also be used for VoIP by connecting them to the Internet via analog telephone adapters, which convert traditional telephone signals into digital data packets that can be transmitted over IP networks.
The broader terms Internet telephony, broadband telephony, and broadband phone service specifically refer to the delivery of voice and other communication services, such as fax, SMS, and voice messaging, over the Internet, in contrast to the traditional public switched telephone network , commonly known as plain old telephone service.
VoIP technology has evolved to integrate with mobile telephony, including Voice over LTE and Voice over NR, enabling seamless voice communication over mobile data networks. These advancements have extended VoIP's role beyond its traditional use in Internet-based applications. It has become a key component of modern mobile infrastructure, as 4G and 5G networks rely entirely on this technology for voice transmission.
Overview
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 used speech coding standards in VoIP are based on the linear predictive coding and modified discrete cosine transform compression methods. Popular codecs include the MDCT-based AAC-LD, the LPC/MDCT-based Opus, the LPC-based SILK, μ-law, A-law versions of G.711, G.722, an open source voice codec known as iLBC, and a codec that uses only 8 kbit/s each way called G.729.
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, 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. 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 phones, VoIP is also available on many personal computers and other Internet access devices. Calls and SMS text messages may be sent via Wi-Fi or the carrier's mobile data network. VoIP provides a framework for consolidation of all modern communications technologies using a single unified communications system.
Integration of VoIP in mobile networks
VoIP technology has been adapted for use in mobile networks, leading to the development of advanced systems designed to support voice communication over modern data infrastructures. Among these are Voice over LTE and Voice over 5G, which enable voice communication over IP-based mobile infrastructures. In contrast to traditional VoIP services, which often function independently of global telephone numbering systems, VoLTE and Vo5G are directly connected to mobile operators' infrastructures, providing seamless connectivity to the international telephone network.VoLTE, introduced as part of 4G LTE networks, enables voice communication over an IP-based infrastructure initially developed for data transmission. It offers features such as high-definition voice and faster call setup times compared to circuit-switched networks.
Vo5G, the 5G equivalent of VoLTE, utilizes the increased speed, reduced latency, and greater capacity of 5G networks to further enhance these capabilities. Both VoLTE and Vo5G maintain compatibility with traditional public switched telephone networks, allowing users to make and receive calls to and from any telephone number worldwide.
These technologies differ from standalone VoIP services by being fully integrated with mobile network operators. This integration ensures additional features such as emergency call support and quality-of-service guarantees, making them a central part of modern mobile telecommunication systems.
Protocols
Voice over IP has been implemented with proprietary protocols and protocols based on open standards 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, which is a connection between two or more peers that provides a context for further communication.
- Signaling – Performing registration and discovery, dialing, negotiating capabilities, and call control.
- Media description – Determining what type of media to send, how to encode/decode it, and how to send/receive it.
- 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, statistics, etc.
- Security – Implementing access control, verifying the identity of other participants, and encrypting data to protect the privacy and integrity of the media contents and/or the control messages.
- Matrix, open standard for online chat, voice over IP, and videotelephony
- Session Initiation Protocol, connection management protocol developed by the IETF
- H.323, 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, connection management for media gateways
- H.248, control protocol for media gateways across a converged internetwork consisting of the traditional PSTN and modern packet networks
- Real-time Transport Protocol, transport protocol for real-time audio and video data
- Real-time Transport Control Protocol, sister protocol for RTP providing stream statistics and status information
- Secure Real-time Transport Protocol, encrypted version of RTP
- Session Description Protocol, a syntax for session initiation and announcement for multi-media communications and WebSocket transports.
- Inter-Asterisk eXchange, protocol used between Asterisk PBX instances
- Extensible Messaging and Presence Protocol, instant messaging, presence information, and contact list maintenance
- Jingle, 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, 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. 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 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 cable modems have this function built in.
- Softphone 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
Smartphones 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 lines installed internationally were VoIP. 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.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. 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.
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 market.
Skype, which originally marketed itself as a service among friends, began to cater to businesses in 2009, providing free-of-charge connections between any users on the Skype network and connecting to and from ordinary PSTN telephones for a charge.