SIP Trunking - What?
By Chris Palermo
Stated simply, SIP Trunking utilizes Voice over Internet Protocol (VoIP) technology for telephony services. SIP (Session Initiated Protocol) enables the digital streaming of voice and data and can best be thought of as a way to cost effectively integrate digital telephony with a traditional telephone system.
In essence, SIP Trunking provides interoperability that enables businesses that have a traditional PBX-based phone system to expand and use VoIP through their internet connectivity. Your PBX will need a SIP-enabled trunk side and some other connectivity devices to enable it, but the components are inexpensive. You will also want a SIP provider that can offer QoS (Quality of Service) to ensure PSTN-equivalent or better performance.
SIP is a good option for organizations wanting to lower their overall telephony costs by moving to VoIP, but whose previous PBX has not yet reached end-of-life. Telephone system expansions are easily managed by integrating SIP, so your new investments are in VoIP and you can begin achieving the associated financial benefits.
You will need to be sure your internet data connection has sufficient bandwidth to handle your SIP calls. With the use of the internet, PSTN gateways, PRI's and BRI's are no longer needed. Through IP, all the processes are simplified. SIP Trunking is easily scalable, which makes other future expansions and moves easier, while maintaining an architecture that is optimized for efficiency.
Those on SIP Trunks, with the appropriate equipment, will also be able to make video calls, enable video conferences and can take advantage of desktop sharing or shared white board capabilities available through Unified Communications.
SIP Trunking is easy to implement and integrate, it has good scalability, enables the transport of both voice and data and reduces costs.
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