Session Initiation Protocol (SIP) is used to signal and control interactive communication sessions. The uses for such sessions include voice, video, chat and instant messaging, as well as interactive games and virtual reality. The SIP protocol is increasingly being used to provide Voice over IP, Presence and Instant Messaging in Next Generation Networks, and being mandated for many new applications, including 3G telephony. SIP can be used to control Internet multimedia conferences, Internet telephone calls and multimedia distribution, in both the core and the periphery of the communications network.
SIP trunks replace the older PSTN (Public Switched Telephone Network) trunks that network operators use to provide dedicated voice calling functionality to enterprises with an on-premises PBX (Public Branch Exchange).
Whereas PSTN trunks use circuit switched network infrastructure to deliver voice calls only, SIP trunks use a broadband IP connection to deliver voice calling based on VoIP (Voice over IP) along with data access and enhanced communications services.
SIP trunking is separated from entirely cloud hosted services like hosted PBX by the requirement for the enterprise customer to have an on-premises PBX with IP interfaces, an all-IP PBX or an IAD (Internet Access Device). SIP trunking is a helpful transitional technology that allows enterprises to benefit from enhanced communications services while continuing to draw value from existing investment in on-premesis equipment.
SIP trunking has many benefits over PSTN trunking, including:
To find out more about how CT CloudSIP trunking could work for your organization, contact CallTower for a demo today: