Voice communications is without a doubt an essential part of just about every business. For many the telephone is the most essential tool. And when a new technology comes along, such as VoIP it takes a while for the idea to catch on. Voice communications is without a doubt an essential part of just about every business. For many the telephone is the most essential tool. Without it they make no sales calls, take no orders, and provide no customer service. When a new technology comes along, such as Voice over IP, aka VoIP, it takes a while for the idea to catch on.

As more businesses see the potential cost savings that VoIP systems make possible, the idea gets even more attractive. Some of the biggest savings come when your company has multiple locations, and you route all voice traffic between offices over your VoIP system. Then things start to get really interesting.

Hardware & Software

One of the first questions you might ask would be what hardware and software will I need to implement a VoIP system? The answer depends on what you want to do. With the convergence of data, voice, and video, the possibilities are pretty broad. The key to making a VoIP solution work is all about QoS. If the phone call doesn't sound as clear as a regular analog phone line, you'll probably get push back from the users.

At the lowest end you can start with a simple IP phone that has a built-in adapter. These types of phones connect directly to your network as opposed to a regular phone outlet.

On the software side there are basically two options. You can go with a traditional vendor and its proprietary software, such as Cisco or Avaya, or you could go with an open-source solution such as Asterisk (www.asterisk.org). Asterisk is a complete PBX (private branch exchange) in software running on top of Linux or Mac OS X. Asterisk supports most of the telephony standards including SIP (Session Initiation Protocol) and H.323. With Asterisk you can build it yourself or turn to a company such as Four Loop Technologies for a turnkey solution.

Standards compliance is important if you want to interoperate with other systems. SIP is an IETF standard for creating, modifying, and terminating sessions with one or more participants. It is widely used as the signaling protocol for most VoIP systems. H.323 is a similar standard from the ITU (International Telecommunication UNI0N) based in Geneva, Switzerland. Many European systems use this standard in lieu of SIP.

Communicating between different systems requires a solution that can translate between the different standards. Asterisk supports both SIP and H.323, so you could initiate a call on a SIP phone and connect with an H.323 phone on the other end. Most of the major VoIP vendors provide support for both protocols in their products.

Avaya

Companies with distributed offices represent one of the biggest potential savings areas for a VoIP implementation. Avaya has several new products that directly address this situation. The Avaya one-X Quick Edition is a SIP-based solution targeted at small businesses and branch offices with fewer than 20 users. Embedding the functionality of an IP PBX into the individual phone eliminates the need for a central server. All you need to set up communications is a PSTN gateway and basic phones.

The Avaya VPNremote for IP Phones is a solution targeted at the teleworker that combines VoIP with VPN functionality into a single device. With the Avaya VPNremote solution you get the same phone experience and network connectivity you'd expect in the office from virtually anywhere. The Avaya VPNremote solution also supports VPN environments from Cisco Systems and Juniper Networks.

Cisco

It's just about impossible to talk about any product that uses IP without mentioning Cisco. Cisco has been delivering voice solutions since before the term VoIP was even mentioned. Cisco Unified Communications represents a wide range of products and solutions that address the needs of delivering voice, video, and IP communications over the same network.

One of the key advantages to a Cisco solution is the breadth of offerings that the company provides. Many organizations already have some type of Cisco equipment in the enterprise networking infrastructure. Adding voice capability in many cases is nothing more than a software upgrade. Having a single vendor providing all your voice and data hardware has both good points and bad.

Cisco has five basic technology components that make up the unified communications system. These are: IP Telephony (call processing, telephones, and endpoint); Unified Communications applications (unified communications clients, messaging, and conferencing solutions); contact center applications (a complete line of products for call center support); communications infrastructure (traditional routers and switches); and management tools (unified operations manager and service monitor).

Cisco's Unified CallManager and Unified CallManager Express products utilize existing Cisco networking products along with Cisco Unified IP phones to deliver enterprise-ready voice solutions. The CallManager Express product offers a low-cost solution suitable for the small business or a branch office.

Cisco's high-end products provide the hardware and software to support the largest call center. Management tools are a key part of all Cisco products, and that includes its Unified Communications offerings. The Unified Operations Manager and Unified Service Monitor tools make it possible to keep high-volume, mission-critical communication centers up and running with a minimal amount of staffing.

Edgewater Networks

The EdgeMarc 4500 series converged networking router provides a host of interconnection options to implement a VoIP solution. Interface options include up to four T1 WAN connections or one Ethernet WAN, a 4-port managed VLAN switch, optional 802.11 wireless access point, and integrated analog phone and line ports.

Flexibility is key to the EdgeMarc 4500 with the ability to function as a low-cost WAN router all the way up to a SIP trunking and enterprise video over IP platform. Another part of flexibility is the ability to reroute calls locally or over the public switched telephone network in case of a WAN link failure.

The EdgeConnect 800PoE managed power over Ethernet switch is a perfect companion to the EdgeMarc device. It provides plug and dial capability to just plug in an IP phone and have it work. It also implements a variety of QoS and VLAN techniques to ensure that voice gets first place in most every case. PoE eliminates the need for additional power supplies and AC wiring with IP phones.

Four Loop Technologies

In most traditional business telephony installations you will find some type of PBX or public branch exchange. Four Loop Technologies has taken the approach of replacing the traditional analog hardware PBX with a software digital one. Switchvox is a complete business phone system in software supporting unlimited extensions and voicemail.

The hardware part of the Switchvox system is essentially a Linux server with the appropriate hardware cards to interface to any existing telephony equipment. You can purchase a T1/E1 interface directly from Four Loop Technologies that will connect your Switchvox server to a traditional PRI or other T1 or E1 service.

Telephone handsets can be either traditional analog devices or SIP digital phones. Analog handset connections to the Switchvox server require an ATA (analog telephone adapter). Each analog handset can have a corresponding ATA, or you can use a channel bank to use existing phone wiring.

Switchvox supports a number of advanced features such as call routing to enable you to receive calls anywhere, voicemail and email integration where your voicemails get delivered to your email inbox, and the ability to create interactive voice response systems using a graphical editor.

by Paul Ferrill