Voice over IP

  • 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 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.[1] 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.[2] voip provides a framework for consolidation of all modern communications technologies using a single unified communications system.[3]

  • pronunciation
  • protocols
  • adoption
  • quality of service
  • performance metrics
  • pstn integration
  • fax support
  • power requirements
  • security
  • caller id
  • hearing aid compatibility
  • operational cost
  • regulatory and legal issues
  • history
  • see also
  • notes
  • references
  • external links

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 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.[1] 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.[2] VoIP provides a framework for consolidation of all modern communications technologies using a single unified communications system.[3]