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SIP contact centers foster integration

Session Initiation Protocol (SIP) can bolster contact center capabilities, allowing integrated communications and customer service. The implementation of SIP, however, can require a special interface and protocol translation.

SIP, the Session Initiation Protocol, has emerged during the last few years as the de facto standard for unified communications and public telephone network access, bolstering the technology contact centers depend on. Contact center architects can leverage SIP as the basis for unified communications at the same time that they expand their contact center capabilities. But a successful SIP deployment requires careful attention to testing, management and interoperability.

Nemertes Research recently reported that 86% of companies are deploying or planning to deploy UC in the contact center in the next two years. But actually implementing UC, which we define as the integration of voice, video, messaging, and conferencing via a common set of user clients, is often easier said than done. In fact, the majority of IT leaders we interviewed cited UC integration and interoperability as their greatest UC challenge.

Here SIP plays a role in providing a common framework for application interconnectivity. SIP's design as an open signaling protocol capable of supporting virtually any media type makes it well suited to enable interconnectivity between related applications.

Contact center architects can leverage SIP as the basis for both application and device integration. For example, SIP-based interconnectivity can allow contact center agents to both see and share presence status to determine subject expert availability and initiate any form of supported communication (such as instant message, voice call, video conference or Web conference). SIP-based automatic call distributors (ACDs) can interface with SIP-based IP PBXs to support call routing and the ability to sign in at any phone or workstation. To make these scenarios work, applications must support SIP, and they must support interoperable SIP primitives (standardized features).

Interoperability in SIP contact centers

This is where things get a little tricky. While SIP provides a framework for interoperability, it leaves many of the details -- such as how codecs and encryption are negotiated, how features are delivered, and how network address translation barriers are overcome -- to application designers. So while vendors may claim they "support SIP," users often find that SIP compliance doesn't mean interoperability.

There are ways around this challenge, however. IT architects and contact center managers can leverage SIP session management to create a SIP-based connectivity layer for existing applications. In this approach, an organization deploys a SIP session manager to provide a presence and signaling layer above existing applications. The SIP session manager interfaces with existing systems via a variety of supported protocols (such as QSIG and H.323) and translates legacy protocols to SIP to enable presence sharing.

For example, a company with a legacy TDM ACD can interface it via QSIG to a SIP session manager to enable call transfer or on-hook/off-hook notification to other applications without having to replace the existing ACD. In this manner, session management provides the benefits of SIP while enabling companies to migrate their legacy contact center infrastructure to IP at their own pace.

SIP contact center softphones have many applications

But UC isn't the only success story for SIP in the contact center. IT leaders can also use SIP to enable a greater variety of endpoints, such as low-cost SIP softphones (with or without USB-based desktop phones) for their remote agents or softphones embedded into existing contact center applications.

In the case of softphones, SIP lends itself to easier support for teleworkers or distributed agents without requiring on-premise ACDs. Some companies are even implementing or trialing SIP softphones embedded into customer-facing websites to support click-to-call directly through a website. In this example, a customer would initiate a call to a support representative through the company's website, using the customer's own PC microphone and speakers (or headset) to speak to the live agent.

This approach offers the opportunity for the agent to speak to the customer and also potentially use additional capabilities such as text chat, screen sharing or video to better respond to customer needs. The company saves money by eliminating costly toll-free inbound calls while it provides increased flexibility and efficiency in customer response.

About the author:

Irwin Lazar is the vice president for communications and collaboration research at Nemertes Research, where he develops and manages research projects, develops cost models, conducts strategic seminars and advises clients. Irwin is responsible for benchmarking the adoption and use of emerging technologies in the enterprise in areas including VoIP, unified communications, video conferencing, social computing, collaboration and advanced network services.

This was last published in October 2010

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