What is SIP Trunking?
SIP, or Session Initiation Protocol, is the standard communications protocol for voice and video in a Unified Communications (UC) solution across a data network. A SIP trunk replaces the need for traditional analog, T1-based Public Switched Telephone Network (PSTN) connections with termination instead provided over a company’s public or private Internet connection through a SIP provider. These SIP providers, often referred to as Internet Telephony Service Providers (ITSP), provide PSTN service on a per minute or channelized pricing model.
The per-minute pricing model is fairly self-explanatory, with a set rate per minute of usage. A channelized pricing model typically provides nearly unlimited minutes on a set number of channels, or call paths. For example, a company can purchase 10 channels and make use of unlimited minutes on those channels, but can only have 10 simultaneous calls.
Many companies already use VoIP within their PBX on the Local Area Network (LAN) to connect to IP phones. SIP Trunking also uses VoIP to take advantage of shared lines, such as a company’s Internet connection, to allow more flexibility in communications. Traditional legacy systems, that aren’t already VoIP-capable, can be connected using common VoIP gateways to take advantage of SIP trunking and reap the significant cost benefits.