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Some protocols perform multiple functions, while others perform only a few and must be used in concert. 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.Ī variety of functions are needed to implement VoIP communication.

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Full words, voice over Internet Protocol, or voice over IP, are sometimes used. VoIP is variously pronounced as an initialism, V-O-I-P, or as an acronym, / v ɔɪ p/ ( VOYP). VoIP provides a framework for consolidation of all modern communications technologies using a single unified communications system. Calls and SMS text messages may be sent via Wi-Fi or the carrier's mobile data network. In addition to VoIP phones, VoIP is also available on many personal computers and other Internet access devices. These solutions typically allow dynamic interconnection between users in any two domains of the Internet, when a user wishes to place a call. Third-generation providers, such as Google Talk, adopted the concept of federated VoIP. This limited the freedom of users to mix-and-match third-party hardware and software.

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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. 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.Įarly providers of voice-over-IP services used business models and offered technical solutions that mirrored the architecture of the legacy telephone network. The most widely used speech coding standards in VoIP are based on the linear predictive coding (LPC) and modified discrete cosine transform (MDCT) compression methods. 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. They transport media streams using special media delivery protocols that encode audio and video with audio codecs and video codecs. 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.

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.