VoIP (voice over IP) is the transmission of multimedia content and voice over Internet Protocol (IP). It is a category of hardware and software which enables the people to use the Internet as the transmission medium for telephone calls. It sends voice data in packets using IP rather than by traditional circuit transmissions of PSTN.

Working of VoIP

VoIP encapsulates audio via a codec into data packets, transmits them across an IP network and encapsulates them back into audio at the other end of the connection. VoIP endpoints include dedicated desktop VoIP phones, softphone applications running on PCs and mobile devices, and WebRTC-enabled browsers.

VoIP enables providers to deliver voice services over their broadband and private networks and allows enterprises to operate a single voice and data network. It also reduces network infrastructure costs and piggy-backs on the resiliency of IP-based networks by enabling redundant communications between endpoints and networks.

VoIP protocols and standards

VoIP endpoints typically use International Telecommunication Union (ITU) standard codecs, such as G.711 or G.729 which are the standard for uncompressed and compressed packets respectively. Many equipment vendors also use their own proprietary codecs. Voice quality may suffer when compression is used, but compression reduces bandwidth requirements. It also supports non-voice communications via the ITU T.38 protocol for sending faxes over a VoIP or IP network in real time.

Once the voice is encapsulated onto IP, it is typically transmitted via the real-time transport protocol or through its encrypted variant, secure real-time transport protocol. The Session Initiation Protocol (SIP) is most often used for signaling that is necessary to create, maintain and end calls. Within enterprise or private networks, quality of service is typically used to prioritize voice traffic over non-latency-sensitive applications to ensure acceptable voice quality.

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