Telephony 101: Giving Voice to Your Network

PBX: The Heart of Voice Traffic

Today's PBX generates the required signaling information for calls outside the organization, passing the connection through a voice trunk and up to the local telco's CO (central office). The obvious advantage is that it lets you avoid intra-office tariffs and reliance on a LEC (local exchange carrier) by aggregating calls into digital trunks. And it's easier and cheaper to manage one PRI or T1 than multiple analog lines from the local CO.

Over time, PBXs have evolved from monstrous electromechanical beasts to refrigerator-sized proprietary boxes to modern rackmounted chassis. Most of these devices run ACD (Automatic Call Distributor) applications that provide advanced call management for inbound calls. ACD apps can direct calls to hunt groups, for example, so if the first line is busy, a call will roll to the next one or to voicemail or a VRU (Voice Response Unit).

With VoIP (Voice over IP) modules now widely available for PBXs, you can upgrade a traditional PBX with VoIP and run your voice traffic over your IP backbone (see "2003 Survivor's Guide to Digital Convergence"). PBX vendors are also offering IP-only PBXs, which means you no longer have to install hardwired extensions when you add new users and handsets.

VRU/IVR: Talking BackVRUs are essentially voice computers that customers use to interact with a company's systems. Just as a browser is your interface to the Web, your phone keypad serves as the user interface to the VRU. Like PBXs, VRUs were once vendor-specific solutions running their own operating systems on proprietary hardware. Today's VRUs rely on standard server hardware and OSs with IVR (Interactive Voice Response) applications, such as those from Syntellect running Windows 2000.

When you reach a business' automated voice system and press "1" for a name directory, you're interfacing with the VRU's IVR application. The IVR guides you to whatever information you're looking for, such as the extension for Joe in accounting.

The ACD transfers calls to an IVR app, which holds the call until the customer either completes the session or opts out by choosing to talk to a human being. The IVR app responds to the caller's queries using its internal data or by connecting to external databases. IVRs can be powerful tools for business--the more calls they can handle, the fewer call-center staffers you need. On the flip side, a poorly written IVR app can cause serious (and potentially costly) customer-service problems.

Advanced IVR applications let users conduct transactions such as bank balance transfers and provide write updates to back-end databases.

CMS: Accounting for Your CallsWith a CMS (Call Management System), administrators can track inbound and outbound call volume, average time on hold, length of calls, IVR opt-outs and other call minutiae. This detailed reporting reveals trends in customer call volume, which is crucial for planning line capacity and staffing and for developing IVR applications. CMS reports may be used to pinpoint inefficient call paths, usually evident when users opt out of the IVR app to talk to a service rep, and detect call volume trends, like the fact that volume is highest at noon on the first Monday of every month.

You can use this information to adjust your work flow and business processes, and update your IVR applications. And management can measure a call-center agent's performance with daily and weekly metrics, such as time to answer, call duration and transfers. Older CMS apps interfaced to PBXs and VRUs over serial connections, but newer ones, such as Avaya's CMS, use IP.

CTI: Caller ID

Computer telephony integration applications automatically pull caller information from the ACD and capture the caller's interactions with the VRU and associated databases. They then send the data to a call-center representative's PC or terminal screen, enabling the agent to greet the caller by name and immediately access that person's account information. The net result: shorter call time, which drives up efficiency at the call center and improves customer service (see "Go With the Flow: How IVR Handles a Customer Call," at left).Using Ethernet and IP connectivity facilitates interdevice communications. But IP connectivity comes with security risks, which result when legacy equipment is added to the mix.

IP enablement isn't the same as VoIP. While VoIP actually puts voice traffic on your data lines, IP connectivity basically allows you to use your existing cable plant for data transfers. IP-enabling your telecom equipment means adding LAN connectivity, so that the CMSs, PBXs and VRUs run on Ethernet, for instance, instead of connecting via conventional serial connections and proprietary cabling schemes. It's simple to do: Older equipment can be upgraded with Ethernet add-ins, and modern equipment already ships Ethernet-ready. But before you plug into your existing switch, there are design and security issues to consider.To simplify configuration and management, you can install your voice equipment on a VLAN or physically separate your voice equipment onto its own LAN segment. This can also reduce your vulnerability from potential holes in your legacy voice equipment. Many large-scale PBXs, including devices from Avaya, Nortel Networks and Siemens, rely on vendor dial-in or VPN RAS for maintenance and upgrades. Direct-dial modem ports made sense at one time, but today they present a back door to your data network. So if you have older telecom equipment, choose a secure remote-access solution instead.

Regardless of your gear's vintage, make sure vendor and telecom-administrator user accounts comply with your company's security and access policies. It's bad enough if a black hat hacker gains access to your PBX or VRU via a forgotten dial-in default account and password. But it's downright disastrous if a hacker exploits that account on an IP-enabled telecom box to gain access to your whole shop. If a hacker uses telnet to gain access to an admin account on your Unix-based VRU, you're wide open.

So before you integrate your legacy voice equipment into your data environment, take some extra design time to put sufficient security precautions into place for all telco gear. If your budget allows, consider adding a hardware voice firewall solution, such as SecureLogix's ETM, to complement your existing data security precautions. (For more information, see "Dial 1-800 Plug Holes" and "SecureLogix Encore").

Pure VoIP

Voice over IP is the next generation of switched telephony (see "A VoIP Wake-Up Call"). It works like this: An analog signal (human voice) is digitized, compressed, wrapped in IP packets and sent off to a destination address (the other end of the conversation). When the packets reach their destination, they're assembled in proper order, decompressed and played back as an analog signal. This must occur in real time and in full duplex, so that both parties can speak over each other and still be heard, for instance.Although sound quality for IP voice has historically been lacking, CODECs and processor speeds have improved to the point where you can now have clear conversations over public data networks. When you add VoIP in your private network, you get the bonus of being able to manage your own line capacities, which results in superb sound quality. SIP (Session Initiation Protocol) has become the standard signaling protocol for VoIP phones. SIP sets up and tears down sessions in two-way IP calls, as well as in multiparty voice and video conferences.

VoIP can offer some serious cost benefits. By installing IP telephony devices in your branch offices, for example, site-to-site calls can be routed over your data lines. This is especially efficient if your data lines carry high batch traffic at night but low volume during the day. Voice calls hit the PBX and then are routed over your data network--a frame relay connection, for instance--rather than going out over a voice trunk. And you bypass the PSTN and its associated charges. (LEC charges vary--VoIP can make sense if you're stuck in a high-tariff region, or if your enterprise has international branches that are already IP-connected with decent data lines.)

You can also set up new offices or build out existing offices by setting up a virtual VoIP PBX at one location to manage call volume for remote sites. It's also easy to add new IP handsets--you can just plug directly into an Ethernet port or connect to a PC and forget about hardwiring an extension.

VoIP makes moves, adds and changes less painful and less expensive. With conventional telephony equipment, you have to map extensions, program special call-handling features and activate phone sets. When users relocate, their phone configurations have to be modified and/or customized. With VoIP modules in PBXs, users can take their customized phone settings with them, even if they're just changing desks. The configuration data is tied to the user rather than a physical extension; the VoIP module just looks for the IP address of the user's phone, rather than an extension mapped to a specific port on the PBX.

Existing PBXs can be retrofitted for hybrid locations, where you keep your existing separate phone infrastructure and use VoIP for adding new hires. So if your call center expands to support 200 new agents, you can install an IP-telephony module into your PBX and run it off the site's CAT 5 Ethernet for the new hires.Can You Hear Me Now?

So stop tuning out the voice equipment running in your organization. Voice and data are fast converging in application design and in LAN and WAN technologies. If telecom responsibilities are being added to your job description, bone up on your voice skills with vendor training, Web tutorials and traditional coursework. And take a close look at both the opportunities and risks of blending the two worlds in your own shop.

Joe Hernick is an IT director with a Fortune 100 firm; he has 12 years of consulting and project management experience in data and telecom environments. Write to him at [email protected].

Post a comment or question on this story.1. A customer calls the 800 number to check the status of a shipment.

2. Based on time of day and/or other rule sets, the call is transferred to an available call center and received by the local PBX.3. The ACD (Automatic Call Distributor) software recognizes the call's DNIS (Dialed Number Identification Service), an 800-number feature that categorizes calls going to the same location for specific purposes.

A sales line or a support line, for instance, determines that the caller should enter a specific IVR (Interactive Voice Response) application running on the local VRU (Voice Response Unit). The call is transferred from the PBX to the IVR app on the VRU via an internal analog port.

4. The customer interacts with the IVR using either touch-tone input or voice-recognition prompts.

5. The IVR app interacts with an external database based on the customer's responses to scripted questions. Then:

(A) If the IVR app satisfies the customer's query, the session ends and the voice ports on the VRU free up for the next caller; or(B) If the IVR couldn't answer the customer's question, the call opts out to a queue for a human agent.

6. As soon as an agent is available, the call is transferred and the agent-to-customer portion of the call commences.

7. With an advanced CTI (computer-telephony integration) system, the call-center agent's PC receives the caller's information (from caller ID and IVR prompting), along with relevant information from the external database.