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SIP trunking is a voice over Internet Protocol (VoIP) technology and streaming media service based on the Session Initiation Protocol (SIP) by which Internet telephony service providers (ITSPs) deliver telephone services and unified communications to customers equipped with SIP-based private branch exchange (IP-PBX) and unified communications facilities. [1]
The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating communication sessions that include voice, video and messaging applications. [1] SIP is used in Internet telephony, in private IP telephone systems, as well as mobile phone calling over LTE . [2]
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A session border controller (SBC) is a network element deployed to protect SIP based voice over Internet Protocol (VoIP) networks. [1]Early deployments of SBCs were focused on the borders between two service provider networks in a peering environment.
A SIP provider (Session Initiation Protocol) is any telecommunications company which provides SIP trunking to customers, usually businesses. Many companies provide SIP "termination" (outbound calling) and "origination" (inbound calling, usually with a plain old telephone service (POTS) phone number, called a direct inward dialing (DID).
ITSPs use a variety of signaling and multimedia protocols, including the Session Initiation Protocol (SIP), the Media Gateway Control Protocol (MGCP), Megaco, and the H.323 protocol. H.323 is one of the earliest VoIP protocols, but its use is declining and it is rarely used for consumer products. [1]
Based on these inherent SIP features it is possible to construct a peer-to-peer network of SIP nodes. In another approach, SIP over P2P, the SIP location service is replaced by conventional P2P overlay networking approach, such as the OverSim framework. [5] In this model, the overlay network is used for service or node discovery and rendezvous.
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