Session Initiation Protocol (SIP) Trunking, or SIP Trunking, is an IP trunk service that can be configured to support both traditional Private Branch Exchanges (PBXs) and IP-PBXs. SIP Trunking can be an IP replacement for traditional Primary Rate Interface (PRI), T1 and analog line PBX interfaces, or can be delivered natively to an IP enabled PBX.
Delivered over FairPoint Communications’ private Ethernet network, SIP Trunking connects a customer’s premises-based PBX to FairPoint Communications’ advanced soft switch network, routing voice calls to the Public Switched Telephone Network (PSTN). SIP Trunking can be provided over the same Ethernet circuit that provides your Internet access, eliminating the need for separate voice and data transport.
In addition to basic voice packetization, SIP Trunking supports all traditional functionality associated with PBX interfaces, like call control, and network-supported features, such as caller ID, CNAM and E911.
Get the value of a converged IP solution:
Lower capital expenditures:
Reduce the strain on your IT staff:
Enhanced features and functionality:
The diagrams below illustrate three configurations for FairPoint Communications’ SIP Trunking.
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