Before SIP trunking, when all phone calls went over the PSTN, businesses would purchase “trunks” – a dedicated line or a bundle of circuits – from their service provider. Today, we have adapted the concept of “trunking” to IP networks, resulting in lower telephony costs and better return on investment (ROI).

A SIP trunk is a service offered by a provider to use SIP to set up communications between an enterprise PBX and the provider. A trunk includes multiple voice sessions – as many as the enterprise needs. While some see SIP as just voice, SIP trunking can also serve as the starting point for the entire breadth of real-time communications possible with the protocol, including Instant Messaging, presence applications, whiteboarding and application sharing.

The potential for a better ROI is a key driver of SIP trunk deployments. However, maximum return on investment can be achieved when you extend VoIP outside of the corporate LAN. In terms of infrastructure purchases, SIP trunks provide an immediate cost-savings. They eliminate the need to purchase costly BRIs, PRIs or PSTN gateways.

The productivity benefits with SIP and SIP trunking are also significant. By extending the SIP capabilities of the corporate network outside the LAN, satellite offices, remote workers and even customers can use VoIP and other forms of real-time communications applications to break down barriers of geography to share ideas and increase productivity.

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.

Equipment based on the SIP protocol – SIP phones, IP-PBXs etc. – have been around for some time. Now that SIP trunks have gained momentum, it has become important to ensure that equipment works together.

Like any application that opens the network to the Internet, SIP trunking deployments have security considerations, but there are ways to maximize enterprise security. One of the most effective techniques is to address SIP security the same way data security is addressed – at the enterprise edge. SIP server and SIP proxy technologies offer maximum control over the flow of SIP traffic, enabling the administrator to ensure correct routing, apply verification and authentication policies and mitigate Denial-of-Service attacks.

Voice quality is not an issue with SIP trunking if proper Quality of Service (QoS) measures are applied, such as over provisioning of links, and prioritization of voice traffic. Reliability is also a moot point. In fact, SIP trunks can be more reliable than the traditional PSTN as a number of failover solutions can be implemented.

The PBX is located on the internal network. The PBX must have a SIP-enabling trunking interface. It can either be an IP-based PBX communicating to all endpoints over IP, or it may just as well be a traditional TDM PBX. The sole requirement is that an interface for SIP trunking connectivity is available.

The Internet provides connectivity to the PSTN for communication with mobile and fixed phones.

The PBX on the LAN connects to the Internet via the enterprise border element. The border element may be a SIP-capable firewall or a SIP-enabling edge device, attached to an existing non-SIP-capable enterprise firewall.

Many enterprises are already using VoIP, however, many are only using it for communication on the enterprise LAN. In this scenario VoIP is only being used as a one-to-one replacement for traditional wireline telephony. For all calls made to the outside of the LAN a PSTN gateway on the enterprise edge is used. These businesses realize a solid return on investment (ROI) just by lowering administrative costs and the costs associated with calls made within the company.

With SIP trunking, the potential for ROI is far greater because SIP trunking takes the idea of VoIP a step further, beyond this LAN application. The full potential for IP communications can be realized only when the communication is taken outside of the corporate LAN.

SIP trunking delivers several benefits: Eliminates costly BRIs (Basic Rate Interfaces) and PRIs (Primary Rate Interfaces, subscriptions, no need to invest in PSTN gateways and additional line cards as you grow. Edge devices offer low investment path in adding new lines as they are typically cheaper per line than the corresponding PSTN gateway. Optimal utilization of bandwidth by delivering both data and voice in the same connection. Maximum flexibility in dimensioning and usage of lines as you avoid having to buy capacity in chunks of 23 (T1) or 30 (E1) lines. Flexible termination of calls to preferred providers; calls to anywhere worldwide can be made for the cost of a local one. Redundancy with multiple service providers and links.

With a SIP trunking solution, the capacity you need when you need it is always available. Instead of dimensioning the telephony for peak usage, it may instead be dimensioned for average usage, allowing the dynamics of QoS to make sure that voice traffic always gets the capacity it needs.

When the capacity of the PSTN gateway and/or PRI connection is reached, it is necessary to invest in an additional PSTN gateway and/or PRI subscription. Unfortunately, this is true even if you only need one more line. Going from one E1/T1 to two always requires additional hardware and they can only be bought in steps of 23/30 lines. Even if you move from an E1/T1 to a higher-level standard bundle like STM-1 the hardware will need to be replaced. The SIP edge device does not have that problem.

In a SIP trunk solution, the enterprise can increase one line at a time by: Purchasing additional software licenses for the edge device. Allocating a greater percentage of the bandwidth for voice. Only if the total bandwidth capacity is used will the Internet connection need to be upgraded.

The use of IP makes it possible to cost efficiently use SIP trunks from multiple service providers, depending on optimal availability and the best rates (capitalizing on time zone differences, and geography.

These routing decisions can be made by the PBX or by the edge device. The fact that this ability can be built into the edge device means that low functionality PBXs can perform routing functionalities as well. By outsourcing this function to the edge device, the PBX needs only to send the number as it is, and let the edge device act depending on destination etc.

Today, calls that could be transferred over IP all the way are connected through TDM connections instead. These situations arise when calls are routed to a PSTN gateway on the LAN. In essence the true benefits of IP communications are not only unrealized, they are defeated as quality will suffer by analog/digital transcoding several times over.