Dial IP for Business

Leon Erlanger

Despite a lot of initial hype from vendors, service providers, and the press, IP telephony (also known as Voice over IP, or VoIP) has been slow to take off. The excitement of using services such as Net2Phone to bypass tolls by sending voice calls over the Internet subsided as charges for traditional long distance plunged. The tradeoff—less-than-perfect voice quality and reduced convenience—was not worth the small savings.

For small businesses, however, VoIP means a lot more than cheap calls. It means getting all the features (and more) of PBXs, key systems, and Centrex services for much less money by creating a phone system using existing in-house data networks. And over the past year, a lot of factors have come together to position IP telephony as a strong contender for small-business communications.

IP phone systems for small and midsize offices have matured and are much more standardized and reliable than they once were. The phones look like traditional handsets, voice quality is excellent, and you get most or all of the features of traditional systems, including call forwarding and transferring, conferencing, message waiting, and voice mail. And now, small businesses can get what were once prohibitively expensive advanced features like call-detail reporting, ACD (automatic call distribution) for service and sales departments, and multiple multilevel auto-attendants—the sophisticated digital receptionists that ask you to press one for Sales, two for Tech Support, and so on.

Bandwidth is cheap. Many small and midsize companies have already upgraded to the switched Fast Ethernet network architecture recommended for LAN-based IP telephony, and today’s affordable high-bandwidth T-1 and DSL wide-area connections make tying in remote offices easier.

The applications that IP phone systems enable are important to businesses that rely on a mobile workforce, e-commerce, and customer relationship management (CRM). Employees working away from the office can connect via the Internet to the company phone system and use most of the system’s features as they would at the office. They can make and receive calls using their office phone numbers. This gives remote workers more company presence. Remote phone-system access is especially useful in customer-service environments with unpredictable peak loads. Supervisors can quickly add off-site operators, transferring calls from a single call-in number.

Implementing the typical CTI (Computer Telephony Integration) functions—linking phone systems with customer database applications and the Web, for example—is much easier when your phone system uses the same IP infrastructure as your data. VoIP enables valuable interactions such as Web click-to-talk, which lets customers browsing your Web site speak to live service reps by clicking a Call Me button and typing their phone numbers.

IP telephony also makes unified messaging relatively simple. You can access voice mail, e-mail, and faxes through a single interface from anywhere via telephone, cell phone, a Windows notebook, or even a PDA. Mobile sales and service staff receive customer requests more quickly.

Combining voice and data into one network can yield significant savings, particularly in new office installations. IP PBX systems are much less proprietary than their mainstream cousins, and their features are accessible with familiar Windows-like interfaces. Moving, adding, and changing users and phones are almost effortless on an IP network. Implementing changes with traditional systems usually requires an expensive visit from the vendor or a consultant. And the integration of VoIP with wireless LANs gives retail, warehouse, and other highly mobile environments new flexibility.

How It Works

IP telephony describes any solution that digitizes voice and transports it over an IP network. Because the VoIP connection is packet-switched, data and multiple voice streams can share a single IP connection.

A phone call requires more than voice transport. The call setup process must produce a dial tone, accept and process a phone number, generate a ring at the other end if appropriate, produce a ring or busy signal at your handset, and connect the call when the receiving party picks up.

The ITU (International Telecommunication Union) and the IETF (Internet Engineering Task Force) have established protocols for simulating call setup and takedown. Most IP telephony products use ITU H.323 or IETF SIP (Session Initiation Protocol). SIP comes from the data networking world and is somewhat lightweight; H.323 comes from the telecom world and is a heavy-duty collection of protocols for call setup and other functions. Each protocol has its supporters, but both standards work, and most vendors make their products interoperable.

After the initial call setup, the actual encoding and exchange of data occur. A software or hardware codec—in the IP handset, the PC, or the server—samples the analog voice signal, converts it to a bitstream, compresses the data, and outputs the result at a certain defined rate. The data is then packetized, sent across the network, reassembled, decompressed, and converted back to an analog voice signal.

The ITU defines a number of standard codecs. Their data rates, levels of compression, and delay characteristics are different, but their names all start with G. They include G.711u (used in North America) and G.711a (used outside North America), G.729, and G.723.1. G.711u is used by the PSTN (Public Switched Telephone Network); G.723.1, G.726, and G.729 tend to be used more in the VoIP world.

Finally, a voice gateway connects the IP telephony network to any other type of network, including a PBX or the PSTN. There are almost always one or more gateways in a VoIP configuration.

Small-Business Voip Architectures

IP telephony is far from an all-or-nothing proposition. You can retain your PBXs and key systems at your various locations but use gateways to translate and send voice calls and faxes over your IP WAN. You save on calls and faxes among telecommuters, road warriors, and disparate sites, especially those overseas. You can choose from a few different types of gateway solutions.

Optional PBX add-in cards that you purchase from your PBX vendor or a third party such as Intel;

Standalone gateway boxes (from third-party vendors such as Cisco, Multitech, and SMC) that sit between your WAN and either your phone system or individual phones (and fax machines);

IP-telephony–enabled routers, router add-ons, FRADs (Frame-Relay Assembler/Disassembler, the equivalent of a router for frame-relay networks), and switches from 3Com, Cisco, Nortel, and other networking vendors.

You can even use a single gateway to extend your PBX services to the other end of the connection, which can be a broadband or dial-up VPN over the Internet. Then remote users can use softphones in their notebooks or special PBX-compatible USB phones to access regular PBX features. Depending on the product, remote workers can even dial extensions and receive calls as if sitting at their office desks.

Moving Up

The next step up is the LAN or IP PBX, which makes particular economic sense if you’re installing a new phone system. A LAN PBX eliminates the traditional PBX, key system, or Centrex service, and uses the LAN for both voice transport and call control. It uses a call-processing and telephony application server and special IP phones that connect directly to the Ethernet LAN.

The server can be a typical Windows NT or 2000 server with special software and telephony cards (to provide connections to analog handsets and phone trunks), or a turnkey server appliance with slots for telephony cards. The 3Com NBX platform, which runs about $14,000 for a setup with 25 IP phones, uses the latter and is popular in the small-office market. IP phones can be pricey (around $300), but their expense is offset by the single cabling infrastructure and the effortless changes that IP allows. Another configuration, used by the Artisoft TeleVantage platform, among others, lets you retain your legacy analog and digital phones and uses the LAN for call control only. Analog voice is encoded on the server, and gateways connect to the PSTN. You can also get complete voice and data solutions from Nortel Networks and Praxon. These systems integrate a PBX, router, VPN, hub, and mail server in a single box.

If you’re looking for something closer to a Centrex service arrangement, you can now get IP Centrex services from GoBeam (in California), SBC Communications, Verizon, and others. These services use IP over a DSL or T-1 to connect you to the providers’ own softswitches (telephony servers), or to typical phone company switches with IP gateways. Such setups provide a complete package of Centrex-style services plus unified messaging for a very reasonable monthly fee and little or no up-front expense. The service either rents you IP phones or installs a rental gateway and router on your premises so you can use your legacy phones.

The IP telephony solution you choose depends on whether you already have an existing phone system and how much of a commitment you want to make to VoIP. A good way to get your feet wet is by using gateways for toll bypass or by trying an IP Centrex service, then taking the IP PBX plunge when you’re comfortable with VoIP technology and convinced of its benefits.

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