Contact Center: Unplugged

Why Does SIP Trunking Matter?

Author:  Jim Mitchell

A few months ago, Aspect Software announced that we had successfully demonstrated interoperability of our product portfolio with BandTel’s Session Initiation Protocol (SIP) Trunking. As I have visited with customers in the recent months, I have received many questions about why SIP Trunking is important, so I thought I would address this topic in my blog.

We are seeing many contact centers moving to SIP-based systems and products as they incorporate Voice over Internet Protocol (VoIP) into their operations.  But, companies that want to use SIP to fully support virtual contact centers in highly-distributed environments need to learn how to manipulate inbound SIP calls between heterogeneous products (e.g., Aspect contact centers, third party contact centers, and IP PBXs).  SIP Trunking is the most inexpensive and efficient way to accomplish this goal. 

Let’s talk about the different ways you can send a contact to the right resource – you can redirect the call or you can use pre-call routing.  Redirecting the call requires that you have tie lines or “take back and transfer” capabilities, both of which cost a significant amount of money when you’re dealing with a high volume of calls and distributed, remote sites.  SIP Trunking, on the other hand, enables you to easily and cost-effectively accomplish pre-call routing. 

Here’s how it works: with SIP Trunking, SIP calls are taken directly from your telecommunications service provider and load balanced across all of your sites or put into a voice portal or other call classification system so that they can be referred to the appropriate resources – a contact center product or agent. There is no limit on the number of times you can direct or redirect a contact without incurring take back and transfer costs, assuming your SIP provider doesn’t charge for it.  Alternatively, you can use your own network infrastructure to direct and redirect contacts. Why is this good?  By referring calls, you are freeing up the initial resources to handle the next contact and you are eliminating your reliance on a computer telephony integration (CTI) interface for load balancing.

You can also use SIP Trunking in lieu of an extensive inter-site network for VoIP, allowing you to utilize your SIP provider as your virtual network.  This is important because it gives you seamless connectivity to the public switched telephone network (PSTN).

I think that SIP Trunking is an extremely important piece of the VoIP puzzle because it can help you to continue to provide a high level of customer service and begin to realize VoIP benefits while migrating from TDM to IP.  I’ll share more with you about the high-level benefits of SIP Trunking in future blogs.  For now, I’d be interested to know if your company is exploring SIP Trunking.  Let me know!

Author: Jim Mitchell
Catergories: Contact Center Technology, Standards, VoIP

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