What is SIP Trunking? SIP Trunking replaces conventional TDM-based carrier interconnection on PRI (T1 or E1) connections with direct interconnection of VoIP communications between Enterprise IP-PBXs and the PSTN via a SIP carrier. It uses the SIP standard for call control and provides the advantage that you can combine all forms of traffic (voice, data, video) on a single connection to the service provider. This is significantly lower cost than separate data and voice connections and provides far greater flexibility. In fact SIP can provide call control for all types of traffic on the connection (though the available bandwidth varies by carrier). Security is provided at the boundary by means of a SIP-aware firewall, plus Call Admission Control (CAC) to provide high-quality voice connections.
SIP trunking has the advantage that SIP trunks come in units of one channel, unlike T1 (23 channels) or E1 (31 channels), which are often more than needed for branch offices. By consolidating all the business requirements in one location (e.g. the data-center) significant savings can be made. In addition, some carriers offer burst-able mode SIP trunks, where, for a small fee, additional capacity can be instantly provided to cover short-term traffic peaks.
The ability to use a common connection for voice, data and video is important — not just for cost savings, but also for deployment of unified communications (UC) solutions. Many legacy services require separation of these media (voice vs. data). UC assumes they are unified. So, using a single pipe makes sense.
SIP trunking is now a well-proven technology, with many carriers offering this service for both domestic and international traffic.
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