SIP Trunk
Over the past two decades communications have largely moved from analog signaling, PBX and related infrastructure to digital technologies leveraging the Internet. The shift is primarily driven by the decreased cost and increased resource efficiency that comes from IP communications, without any loss of security or quality. As communications services have converged onto the Internet the ability to integrate numerous services has developed. The result is Unified Communications. SIP Trunking is the delivery mechanism that underpins unified communications, providing the function of connecting an organization’s PBX(s) to the necessary service providers. In doing so, it ensures phone calls and other IP-based communications originating from an employees’ phone or computer are connected to the destination. Enterprises interested in implementing unified communication solutions consider video, presence, instant messaging and other communication and collaboration tools. Voice over IP, a core element of IP communications, is powered by SIP Trunking. Any enterprise looking to implement a VoIP or Unified Communications service must have a SIP Trunking provider.
VoiceTrunking provides SIP Trunk service for Asterisk SIP, 3CX, Aastra, Allworx, AsteriskNow, Cisco, Cisco UC500, Cisco UC520, Cisco UC540, Cisco UC560, Elastix, Epygi, Fonality PBXtra, FreePBX, Grandstream, PBX in a Flash, SwitchVox, Talkswitch, Trixbox and virtually any other SIP PBX
SIP Trunk is a service to route concurrent phone calls over the IP backbone of a carrier using Voice over IP technology. SIP Trunks are used in conjunction with an IP-PBX systems and are thought of as replacements for traditional PRI or analog circuits.
The popularity of SIP Trunks is due primarily to the cost savings of SIP, along with the increased reliability as backed by the SLAs of SIP Trunk Providers. SIP Trunks are cheaper than analog circuits while maintaining the same service quality that businesses expect from line quality. Voicetrunking SIP Trunks cost $0 per trunk, versus $40 per analog circuit. In addition, long distance termination charges associated with SIP Trunks are much cheaper than traditional analog or TDM rates usually under 2 cents per call. When coupled with line oversubscription - for example a 20 person company purchasing just as many SIP trunks as they anticipate having concurrent calls, typically 4 to 6 - SIP Trunks are a very cost effective way for a business to save money. Lastly, a primary benefit of SIP Trunks over PRIs is that SIP Trunks can be purchased in increments of 1, whereas PRIs have to be purchased in increments of 23 channels
A business should consider purchasing SIP Trunks when they decide that their needs are best met with a premise-based PBX system. This system is often referred to as an IP PBX system. Coupled with SIP Trunks, an IP PBX system serves up similar features to hosted solutions. SIP Trunks typically save a business customer money over a hosted solution in that one SIP Trunk can serve the needs of three to four employees while in a hosted PBX solution one hosted seat is needed for each employee.
SIP Trunks are virtual circuits delivered over an Internet Access line so depending on the number of SIP Trunks purchased, and the amount of excess Internet connectivity, a business should consider purchasing more Internet Access. However, it's important to know that when a SIP Trunk is not being used, the bandwidth otherwise allocated to a SIP Trunk is freed up for use in less intensive applications, such as e-mail and general web use. This dynamic allocation of bandwidth is yet another feature of SIP Trunks versus more traditional technologies, such as analog or PRI circuits.
SIP Trunks can work with any PBX supporting SIP protocol. Asterisk has become one of the most popular IP PBX's of the world due to its free, open source licensing, open design, extensibility, and excellent feature set with Asterisk SIP Trunk services. SIP Trunks can also be made to work with traditional analog or key systems with an Integrated Access Device (IAD). The SIP Trunk service provider will need to interoperate with the underlying equipment manufacturer. However, it should be noted that with the advent of standards around RFC 3261 and SIP Connect, the challenge of finding SIP Trunk Service Providers with SIP Trunk compatible equipment is significantly decreased.
FreePBX is an easy to use GUI (graphical user interface) that controls and manages Asterisk, the world's most popular open source telephony engine software. FreePBX has been developed and hardened by thousands of volunteers over tens of thousands man hours. 
