SIP Trunking line converges voice and data onto one pipe, providing seamless access between the Internet and the worldwide PSTN (Public-Switched Telephone Network), resulting in immediate cost saving.
With SIP Trunking the IP Network provides PSTN connectivity. A call travels the majority of its path over the network instead of on the PSTN and then drops back down to the destination/facility at the last mile. By implementing as a carrier, traditional local and long distance charges decrease dramatically because the call travels mostly over the IP network and not on the PSTN.
There are three components necessary to successfully deploy SIP trunks: a PBX with a SIP-enabled trunk side, an enterprise edge device understanding SIP and an Internet telephony or SIP trunking service provider.
In most cases the PBX is an IP-based PBX, communicating with all endpoints over IP, but it may just as well be a traditional digital or analog PBX. The sole requirement is that an interface for SIP trunking connectivity is available.
The PBX on the LAN connects to the ITSP via the enterprise border element. The enterprise edge component can either be a firewall with complete support for SIP or an edge device connected to the firewall, handling the traversal of the SIP traffic.
On the Internet, the ITSP (Internet Telephone Service Provider) provides connectivity to the PSTN (Public Switched Telephone Network) for communication with mobile and fixed phones.
While everyone can agree that a great benefit of SIP Trunking is a typical cost that is 40 – 70% below what legacy Telcos can offer, the features that enhance functionality and scalability and resolve issues related to large-scale enterprise deployments are the major features associated with SIP Trunking. Some of the features of the product line include:
