VoIP (voice over IP) is the transmission of voice and multimedia content over Internet Protocol (IP) networks. VoIP historically referred to using IP to connect private branch exchanges (PBXs), but the term is now used interchangeably with IP telephony.
VoIP is enabled by a group of technologies and methodologies used to deliver voice communications over the internet, enterprise local area networks or wide area networks. VoIP endpoints include dedicated desktop VoIP phones, softphone applications running on PCs and mobile devices, and WebRTC-enabled browsers.
How does VoIP work?
VoIP uses codecs to encapsulate audio into data packets, transmit the packets across an IP network and unencapsulate the packets back into audio at the other end of the connection. By eliminating the use of circuit-switched networks for voice, VoIP reduces network infrastructure costs, enables providers to deliver voice services over their broadband and private networks and allows enterprises to operate a single voice and data network. VoIP also piggy-backs on the resiliency of IP-based networks by enabling fast failover around outages and redundant communications between endpoints and networks.
VoIP protocols and standards
VoIP endpoints typically use International Telecommunication Union (ITU) standard codecs, such as G.711, which is the standard for transmitting uncompressed packets, or G.729, which is the standard for compressed packets. Many equipment vendors also use their own proprietary codecs. Voice quality may suffer when compression is used, but compression reduces bandwidth requirements. VoIP typically supports non-voice communications via the ITU T.38 protocol for sending faxes over a VoIP or IP network in real time.
Once voice is encapsulated onto IP, it is typically transmitted with the real-time transport protocol (RTP) 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 (QoS) is typically used to prioritize voice traffic over non-latency-sensitive applications to ensure acceptable voice quality.
Additional components of a typical VoIP system include the following: an IP-PBX to manage user phone numbers; devices; features and clients; gateways to connect networks and provide failover or local survivability in the event of a network outage; and session border controllers to provide security, call-policy management and network connections. A VoIP system can also include location-tracking databases for E911 (enhanced 911) call routing, and management platforms to collect call-performance statistics for reactive and proactive voice-quality management.
Products migrate from legacy PBX to VoIP
VoIP products can accommodate thousands of users. Discover how Cisco UCM offers VoIP technology that can replace a telephone system. Also get more information on the Panasonic VoIP KX-TDE system, which is designed to support 1,000 extensions and up to 64 SIP trunks; the NEC VoIP product lineup, which consists of the Univerge series; and the ShoreTel VoIP phone system; Mitel 3300 VoIP platform, which supports both small and large enterprises; and the Adtran NetVanta platform.