Call centers taking VoIP to the next level
By Frederic Dickey, VP of Product Management, Marketing and Services at Sangoma
In many corners of the call center world, there is the ongoing challenge of meeting peaks and valleys in demand. Whether it’s an enterprise, multi-location facility, or a smaller scale business with team members working from home offices, always-on connectivity is a prime area of concern – and cost – for call center operations.
Over the past decade, call centers have put significant investment into Voice over IP (VoIP) to integrate a myriad of functions within the call center suite, from inbound and outbound calling, to integrated voice response, analytics and workforce management.
Many of those same operations however continue to rely on traditional telephone lines from local telecom providers for their infrastructure needs. To have VoIP-based call center functions, they must implement VoIP gateway devices that will convert analog traffic (that is, the voice that runs over the telephone lines) to digital (data packets running over the Internet). This lends itself to some significant challenges when it comes to scaling up and down to meet seasonal or daily fluctuations in demands.
At the heart of the problem is the fact that this model often requires a number of telephone lines, which can take weeks if not months before it is operational. Each installation can cost several thousands of dollars, along with a sizeable monthly fee from the provider. In cases where there are multiple call center operations, individual sites may need to work with different providers, depending on their location and available services. This adds to the infrastructure, management, and monthly telecommunications costs.
To put this in perspective, a telephone line can handle 24 simultaneous calls, which means larger call centers may literally have hundreds of lines. This also means large-scale gateway investments to meet that volume. When traffic increases, the only way to increase capacity is to buy and install additional lines and gateways, adding to infrastructure costs and wait times. To mitigate the problem, call centers have tended to over-invest in infrastructure to create buffers beyond current peak numbers.
Another issue is the growing need to use virtual agents in remote locations, including in many cases, a home office. Before VoIP, setup of those agents required special gear including an application server and modems. In addition, a technician would have to go to the agent’s location to perform the initial set up.
Even with VoIP there are challenges in connecting remote agents. Typically the procedure is to set up a virtual private network (VPN) between a home agent and the VoIP call center suite. Technicians would need to manage the VPN server and each home agent would have their own software.
A much more effective and increasingly popular option is moving from traditional telephone lines to a SIP (session initiation protocol) trunk from an Internet telephony service provider. In simple terms, it’s akin to connecting your company’s Internet to the local service provider’s Internet rather than multiple telephone lines. Each connection would come with hundreds and hundreds of megabits of bandwidth that are paid by subscription. In situations where more bandwidth is needed, companies simply contact their provider to enable more ports, and that extra capacity can be added virtually instantly without having to change the infrastructure. Not a single cable will need to be laid.
Moving to a SIP trunk translates into significant reductions in operating expenses, while enabling call center operations to ride the peaks and valleys at less cost and inconvenience. An added advantage over legacy systems is that rather than having to purchase local phone numbers and install telephone lines at every location, businesses can consolidate management into one place, using one provider and deploying one set of phone numbers, because phones are not tied to physical phone lines. Managers simply assign phone numbers, route them over the SIP trunk, and install session border controllers at strategic locations to route them. A session border controller is a hardware device or software application that manages how VoIP calls are initiated, conducted and terminated. Each can securely manage up to 200 simultaneous calls.
Enabling full-, part-time or ad hoc agents at home requires a simple software installation on the agent’s computer, a headset, and a desktop SIP phone. The remote agent then simply registers to the VoIP Call Center Suite to appear in the queue like any other agent sitting at their desk. Everything is integrated and secure.
The benefits of combining VoIP and SIP trunking is best illustrated by a working example. New York-based auto parts retailer Parts Authority operates in more than 70 locations in Georgia, New Jersey, Maryland, DC, and Arizona that collectively handle thousands of incoming calls from customersacross the country.
Over 40 years of doing business, Parts Authority had expanded significantly through organic growth and acquisitions. With that success, the phone system that was in place had become unwieldy and complicated to manage. In fact, 80% of its 70 locations had services with local providers. Parts Authority realized it needed to consolidate telecom expenses, improve response times and call quality, and lay the groundwork for more expansion.
In 2011 the company converted its traditional phone system to a VoIP model. Over time it also decided to centralize management using session border controllers and software. While there are a number of technical aspects involved that are best left to IT professionals, here is a low level explanation of how it works: An incoming call is sent over an assigned SIP trunk via a session border controller to the primary PBX (private branch exchange) for that region. If the SIP trunk is down for any reason, the session border controller automatically routes the call over the backup SIP trunk to the PBX.
If for any reason the primary PBX is down, the backup session border controller will route all incoming calls to the backup PBX. (In more traditional situations, users have to register to a backup PBX in the event of a failure which can take time.) Once service is restored, calls are automatically routed back through the primary SIP trunk to the primary PBX, creating a completely automated failover.
Today all Parts Authority call center locations are on a network with SIP trunks into four regional locations – a far cry from the 50 they were running previously. In doing so it has reduced its reliance on multiple providers to two carriers. In addition, dropped/missed call rates has been reduced from three to five per cent to under one percent; and the man hours to support and maintain the VOIP system have also dropped dramatically because there are fewer PBXs to manage.
To date the number of call centers using SIP trunking is relatively small. But as capacity and flexibility needs grow and technology evolves, it is rapidly becoming the option of choice. Until recently, operations were not even familiar with what a SIP trunk is. Now they are beyond that education stage, and becoming increasingly aware of the business benefits that come with combining VoIP and SIP trunking. With the increasing pressure on call centers to improve functionality and response times, combined with growing demands for remote connectivity, there is already a growing contingent of early adopters.
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About the Author:
Frederic Dickey is the VP of Product Management, Marketing and Services at Sangoma (
sangoma.com), a leading provider of hardware and software components that enable or enhance IP communications systems for both voice and data. Frederic has more than 20 years’ experience in the telecommunications industry. He is responsible for product life cycle management and roadmap development, and leads the company’s online marketing strategies and operations. He holds a Bachelor’s degree in Electrical Engineering (Honours) from McGill University in Montreal.