Voice Over IP Technology Review Research Paper

Introduction

Voice over Internet Protocol (VoIP) technology is a method of providing telephone calls over broadband data connections that provides a number of benefits to both corporate and consumer customers. The service is significantly cheaper than traditional phone service, and it can also support advanced value-added services, such as unified messaging. This tutorial provides an overview of how VoIP works and the benefits it offers.

Voice over IP (VoIP) is one of the most revolutionary telecom services to come about in the last decade. (Beardsley, 2004) What started out as a hobbyists invention to make inexpensive telephone calls over the Internet has evolved into a viable form of business communications. It allows organizations to converge their voice and data networks into one next-generation pipe, delivering substantial cost savings as well amazing flexibility. It can support conference calls, unified communications, advanced routing, and several other types of value-added features. Organizations who use VoIP or are considering deploying the service are faced with several challenges, such as security and quality of service.

Voice over Internet Protocol (VoIP) was first demonstrated in the early 1980s when Bolt Beranek and Newman (BBN), in Cambridge, Massachusetts, set up a voice funnel to communicate with team members on the West Coast as part of their work with the Advanced Research Projects Agency (ARPA). (Chertok, 2004).

The TCP/IP protocol was developed originally by ARPANET as a means for linking disparate data networks, systems, and hardware. In the case of VoIP, the voice funnel digitized voice, arranged the resulting bits into packets, and sent them over the Internet. Additional development of the technology had to wait until the 1990s, when improvements in microprocessors, digital signal processors (DSPs), codec technology, and routing protocols all came together to make feasible products and services for consumer and business use (Hersent, 2004).

IP telephony has developed rapidly to become a successful consumer service offered by major telcos, such as AT&T, Qwest Communications, and Verizon; major cable operators, such as Time Warner Cable, Comcast Cable, and Cox Communications; as well as numerous smaller companies like Vonage, Packet8, and Net2Phone, which specialize in IP telephone service. (Snyder, 2004) Market growth for residential and small office/home office (SOHO) VoIP has been exponential since 2004. Most major providers entered this space because of their ability to deliver a low-cost option to plain old telephone service (POTS) with long-distance carriers. Residential and SOHO customers responded more quickly to VoIP because they tend to be slightly more tolerant of dropped calls or packets and latency (Beardsley, 2004).

Enterprise adoption of VoIP has been slower, however, because POTS has guaranteed a high quality of service (QoS). VoIP technology, however, has begun to close the gap in performance with traditional phone service. Carrier-class gateway platforms that connect to legacy systems via T1/E1 or analog interfaces have matured to reliably handle call management, billing, and authentication. (Hersent, 2004) Service providers have emerged to handle all Operations and Support Systems (OSSs) for VoIP providers that do not have a capable OSS in place, such as some Multiple Service Operators (MSOs) and Internet Service Providers (ISPs). Even major telcos, like Verizon, which launched its service in July 2004, use an OSS service provider to handle the billing and necessary back-office functions. VoIP service providers and customers can now access call detail and billing history and update account information over a secure website in real time.

VoIP Equipment

VoIP implementations require additional equipment compared with traditional public-switched telephone network (PSTN) traffic. As voice is analog, it must be converted into a digital signal. DSPs handle this function and arrange the digitized voice into packets, which are compressed and use less bandwidth than the traditional circuit-based network. Three techniques, or algorithms, that are used for compressing signals are Conjugate Structure-Algebraic Code Excited Linear Prediction (CS-ACELP), Adaptive Differential Pulse Code Modulation (ADPCM), and Low Delay-Code Excited Linear Prediction (LD-CELP). (Blosser, 2003) Codecs, acting as intelligent routers embedded in a VoIP gateway, run the voice data through the compression algorithm, allowing for the signal to be contained in an IP packet. When a signal is received, the signal in the packet is decompressed; however, because of the nature of IP, VoIP calls have been known for latency and lost packets. When making a call, a customer will dial a number that is routed from the PBX to a VoIP service provider. (Chertok, 2004) If network congestion reduces the quality of a call because of latency, intelligent routers can send the call over PSTN rather than have the gateway compress the signal, which becomes less important as IP networks offer a QoS feature that allows voice and video traffic to be prioritized by setting a type of service byte in the header of each packet. Priority is given to voice and video, which are most affected by latency. (Hersent, 2004)

IP platforms are IP gateways that sit between the PBX and the network router and handle the transfer of digital content between various networks, such as PSTN, Asynchronous Transfer Mode (ATM), IP, and Ethernet. IP gateways detect incoming calls, determine the nature of the call (voice, fax, data, or video), determine the call’s intended destination, and pass the signal through in the appropriate manner using the appropriate protocol.

Gateways include software that allows for control of network settings and IP/PSTN interconnectivity and often provides a measure of monitoring and control of the system’s use. Tracking call patterns, for example, and supervision of bandwidth usage can be accomplished from a workstation associated with the gateway to allow additional tweaking after installation to optimize network performance. If the gateway is used to route outgoing calls from the network to the PBX, additional control will be required to detect calls–usually identified by the dialing of a separate prefix–that are off the network and need to be routed through the PBX to the PSTN. Other features, such as least-cost-routing, will allow the system to automatically detect an outgoing call’s destination and select the most appropriate available network path (LAN, WAN, Internet, or PSTN). (Davidson, 2007)

Softswitches are large-scale central connecting and switching devices that are more software-based than its predecessors. Softswitches comprise media gateways and call agents. Media gateways execute the functions of an IP gateway described above. Call agents handle advanced feature sets like billing, call routing, and signaling.

In today’s market, IP gateways tend to be low-end solutions with some drawbacks. The interoperability with a circuit-switched PBX can be inefficient. IP gateways working in conjunction with network address translation (NAT) as a security feature can prevent point-to-point VoIP sessions. (Snyder, 2004) Gateways are inefficient when it comes to compressing or multiplexing packets, which can hurt quality and latency, particularly as call volume or network traffic increases. Softswitches are the vehicle of choice for most major telco operations when replacing legacy switching systems. They enable a broader service offering without the huge hardware capital expenditure.

More recently, IP PBX systems have come on the market led by Cisco Systems and Avaya. These IP PBXs can handle IP voice traffic natively, negating the need for the IP gateway on the local level and irregardless of a softswitch on the network level. IP PBXs are improving in scalability and have been used primarily by small- and medium-sized businesses. Larger organizations, however, are beginning to deploy IP PBXs as enterprise solutions, which have shortcomings of their own, including more complicated implementation, limited failover, and no multiplexing capabilities. Existing circuit-switched PBXs now have the ability to be upgraded with VoIP modules to handle IP traffic, but these act as the IP gateway, which adds to processing time for the voice traffic, and thus increases latency. Cisco, for example, offers IP Communications products for SOHOs, medium-sized businesses, enterprises, and service providers, offering everything from VoIP phones to networking equipment to unified communications-enabled call centers. (Blosser, 2003)

VoIP is now a proven technology with many inherent benefits. As a result, users are looking for such capabilities as IP Centrex and more complex features, such as unified voice, e-mail, and fax messaging. These and other innovations can be implemented quickly and economically through a flexible softswitch architecture that uses workstations to control data calls on IP networks much like circuit switches do for voice calls on the PSTN. In addition, softswitches provide a full range of IP-based communications services that are virtually indistinguishable in quality and ease-of-use from services on traditional circuit-switched voice networks.

Despite concerns about security, lapses in interoperability, and the difficulty in monitoring IP telephony performance standards, which makes it less attractive to go with a VoIP solution for corporate clientele, momentum continues to build for VoIP. Carriers have generally accepted that circuit-switched networks will someday be replaced with packet-switched IP and are gradually implementing and upgrading their infrastructures to handle it for fear of being left behind in this new age of voice-data convergence.

VoIP Services

Companies interested in implementing VoIP may reap the following benefits:

• Flexibility–Voice over IP can support several types of value-added services, such as call forwarding, caller ID, and conference calling.
• Cost–VoIP service is significantly less expensive than traditional voice services.
• Scalability–Companies can easily add additional users to a VoIP network or move employees to a different location without needing to change their telephone extension. (Blosser, 2003)

Current View

As with any relatively new technology, problems exist that must be addressed by equipment manufacturers and service providers alike. How these problems are addressed will determine how fast VoIP can replace circuit-switched voice services, particularly in the enterprise environment.

Call Volume

Before 1998, the market for Internet telephony service had been hampered by equipment that could not handle call volumes comparable with what traditional carriers offered through their circuit switches. In 1998, all major telephone switch manufacturers began offering highly scalable IP telephony gateways that were fully H.323 compliant and supported audio compression methods, including G.723.1, G.726, G.729A, G.729B, and G.711. In addition, they started supporting standard circuit-based protocols, such as SS7, ISDN, PRI, CAS, and C7, as well as packet-based protocols, such as H.323, SIP, SIP-T, and MGCP. They also included full OAM&P capabilities, including carrier billing, storage, and file transfer (Chertok, 2004).

Today’s IP switches closely match the capabilities of legacy Class 4 and Class 5 telephony switches. The GSX family of Open Services Switches and Network Border Switches from Sonus Networks (GSX9000 and GSX4000), (Snyder, 2004) for example, was designed and developed to meet the rigorous requirements of public network service providers, including complete redundancy of all system elements, toll-quality voice, interoperability with the SS7 network (accomplished via the Sonus Networks SGX2000 SS7 Signaling Gateway), scalability to 100 million ports, and compliance with Network Equipment Building System (NEBS) specifications. A single GSX9000 shelf can support more than 8000 simultaneous calls over IP, and a fully configured system can support hundreds of thousands of calls (Richard, 2005).

Latency

The key variable in call quality between PSTN and VoIP is caused by round-trip latency (twice the one-way latency, which is the delay that occurs within the system from the time one person speaks until the other person hears their speech). Round-trip latency for domestic calls over PSTN is usually less than 150 milliseconds (ms). (Davidson, 2007) The standard for acceptable, high-quality voice services call for round-trip latency of less than 450 ms, with anything above 900 ms considered so poor as to be impractical to hold a conversation. It is not uncommon, however, to have up to 5000 ms (5 s) or more of delay on the public Internet with a standard dial-up connection, depending on the distance of the call and the amount of traffic traversing the Internet at any given time. (Kubher, 2004) The Internet, however, does not handle most commercial IP telephony services. Instead, managed high-capacity backbones handle the long-haul portion of the call, greatly reducing delay. For example, AT&T routes its entire volume of CallVantage service over its IP backbone and uses broadband connections to the home (DSL or cable modem) to eliminate most latency. (Shepard, 2005) Even smaller vendors such as Skype signed network deals with backbone providers to carry the service. As a result, only a few true public Internet carriers remain. The large network providers commit to a latency performance standard as part of their Service Level Agreement (SLA). For example, Qwest’s performance is measured against a 50-ms one-way latency benchmark. With ITU standards calling for 300-ms total round-trip latency, VoIP users must have one-way latency of less than 100 ms between their internal network as well as that of their ISP and the ISP and network of the other calling party. (Blosser, 2003)

With satellite communications, latency remains a major concern. As geostationary satellites have an altitude of approximately 36,000 km, with the speed of light constant at almost 300,000 km/s, the optimal round-trip latency just for the earth-to-satellite-to-earth portion of the network is 240 ms. As a result, the standard round-trip latency for satellite VoIP calls is around 500 ms (Chertok, 2004).

Several technologies and techniques are employed to maintain consistent call quality over IP networks, including the use of native IP switches, ATM on high-capacity fiber backbones, and routing protocols that give preference to real-time traffic. Although delay is not entirely eliminated, it is not a prohibitive problem to offering the service. As managed-IP backbones branched out to reach major metropolitan areas, delay has become less of an issue to the point where consumers and corporate accounts are largely unaware that their calls–yes, even a portion of most PSTN calls–are being carried over a packet network.

Voice Quality

Although the poor voice quality offered by the first generation of Internet telephony products condemned them to hobby status, voice quality over IP has continued to improve by leaps and bounds. The use of server-based gateways rather than sound cards in users’ computers provides the processing power needed to minimize compression/decompression time, whereas improvements in DSPs provide high-quality sound. (Davidson, 2007) The mean opinion score used to rate the quality of speech codecs gives toll-quality voice a top score of 4.0. The voice-compression algorithm used in IP telephony applications, known as G.723.1, is rated at 3.98, with scores ranging from 3.57 under network stress to 4.08 under ideal conditions. (Snyder, 2004) Now that the industry has seemingly coalesced around the international G.723.1 standard for VoIP networks, the added benefit of interoperability exists among the products of different vendors (Kubher, 2004).

Ease of Migration

For carrier and enterprise environments, a key issue is how to migrate from traditional circuit-switched technologies to packet-based technologies that can more economically support voice traffic. From the carrier perspective, vendors like Nortel have addressed this problem with solutions that allow a smooth migration to IP according to customer demand. (Richard, 2005) The company’s Succession solutions support both the new IP-based services and the thousands of existing telephony features, such as three-way calling, follow me, and caller ID on a single multi-service packet network, avoiding the requirement for expensive overlay networks. Available Succession solutions include long distance, local tandem, and local line, as well as CATV, broadband wireless, and other last-mile technologies. Succession also accommodates existing circuit switches from other vendors onto the same packet network (Hersent, 2004).

From the enterprise perspective, VoIP solutions can be phased in as legacy PBXs reach capacity. Instead of paying an exorbitant fee for a forklift upgrade, adjunct IP switches can be installed at a significantly lower cost to handle overflow traffic or traffic destined for on-net corporate locations. New branch offices can be equipped with VoIP systems, and telecommuters can be equipped with devices that support VoIP over a dial-up virtual private network (VPN) connection or dedicated broadband connection. The key to an economical migration to VoIP is to leverage the installed base of equipment while transitioning incrementally to VoIP solutions as opportunities occur (Chertok, 2004).

Increasingly, PBX vendors are recognizing the importance of providing customers with easier and more economical ways of supporting VoIP and, in the process, helping them transition to voice and data convergence. For example, Avaya’s DEFINITY portfolio provides enterprises the flexibility of carrying voice, video, and data traffic over the Internet, intranets, extranets, public-switched networks, and ATM. DEFINITY IP Solutions operates both as an IP gateway and as a gatekeeper. As a gateway, it converts voice traffic to IP packets for reliable transmission over the IP network. Its QoS capability monitors the performance of each call. If at any time the IP network’s performance is not acceptable for voice or fax calls, DEFINITY will reroute the call over an alternative network, if available (Davidson, 2007).

Remote User Support

Enterprises with many telecommuters and mobile professionals should consider a VoIP solution that supports remote log-in so these employees can have the same capabilities as their desktop telephone sets, including voice mail access, hold, call forward, transfer, speed-dial, and conference, as well as multiple call appearances and call displays on their laptop computers while working remotely and retaining the same phone number regardless of physical location (Richard, 2005).

This solution can even be applied to call center operations. Avaya’s IP Agent R7, a customer relationship management (CRM) software solution, helps enterprises take full advantage of their data networks by delivering call signaling and contact center telephony features, including IM, screen pops, and interconnectivity for click-to-dial applications, to an agent’s PC through an IP connection. This solution allows agents working at home to provide the same high level of customer care that they can in a traditional call center environment. For the company, the solution can lower the cost of support operations. For a discussion concerning the benefits of using call center agents who work from home, please see the Faulkner article entitled “Home Sourcing: The Virtual Call Center” (Khasnabish, 2003).

Enterprise Service

Companies can deploy VoIP solutions to leverage investments in their corporate intranets without having to rely on an ISP or carrier. Nortel and Lucent offer IP adjunct switches that interface with their PBXs and ACDs. Numerous smaller companies offer IP-PSTN gateways for corporate use as well. Cisco Systems offers an IP PBX solution called the Cisco EGW 2200 Enterprise Gateway, a call processing system that transports intra-office calls over Ethernet and wide-area calls over the PSTN or a managed IP network. (Shepard, 2005) The Cisco family of IP phones offer a wide range of button options and screen, display, and speakerphone configurations. All the phones support G.711 mu-law or G.723 voice encoding. Many of the newer models support power over Ethernet (PoE) standards (Chertok, 2004).

Device installation automation allows an administrator to plug a phone into the IP network, have the device automatically acquire an IP address, register with the system’s CallManager, and download a configuration template and available directory number. The IP PBX can be accessed through a Web browser from anywhere in the world for remote management and diagnostics. The system comes with an interface to the Windows NT Event Viewer, enabling network managers to view system events. Call detail records are available in a flat-file, comma-delimited format that can be imported into other software programs for reporting purposes.

It is not necessary to buy all new phone equipment to implement VoIP. Purchasing a VoIP gateway allows a company to keep its existing phone gear while achieving VoIP’s toll bypass benefits. These systems connect to a PBX’s station or trunk ports. They convert analog voice to digital, and then transmit the signal via a 10BaseT port as IP packets to a private data network or the public Internet. (Khasnabish, 2003) A companion VoIP gateway at a remote site turns voice packets back to a voice signal so it will be accepted by the phone system or directly connected phone. For businesses with high enough intra-company call volume, the savings in long-distance toll charges will typically pay for the gateways in under a year. Enterprises can also purchase IP phones before migrating to an IP PBX based on proprietary technologies or open standards, such as MGCP, which can support business IP telephony systems. After choosing IP phones, the IP PBX, using H.323, SIP, MGCP, or a similar standard, can connect to the VoIP service provider; in which case, enterprise users would not need a local gateway (Davidson, 2007).

Classes of Service

Through a management application, an IP telephone system or commercial service could be configured to assign users a class of service (CoS), just as users are assigned a CoS through a conventional PBX or telephone service. In the corporate environment, certain users, such as clerical staff, might be assigned a class of service that limits them to making calls over the corporate intranet, for example, whereas higher level staff might be assigned a CoS that allows them to use another type of service when the IP telephone system has no ports available.

Managers and executives might be assigned a CoS that permits calls over any type of service, regardless of whether IP ports are available. Different call features can also be assigned to a CoS. For example, if clerical staff have no need for voice mail or conferencing features, their CoS would not allow them to access these features. In similar fashion, a commercial VoIP service provider also can assign subscribers a CoS based on the features and options they want.

Standards

Proprietary VoIP products have given way to those that adhere to the H.323 umbrella recommendation from the International Telecommunication Union (ITU) or the Session Initiation Protocol (SIP) from the Internet Engineering Task Force (IETF). Both H.323 and SIP define mechanisms for call routing, call signaling, capabilities exchange, media control, and supplementary services. For distributed VoIP systems, Media Gateway Control Protocols MGCP and Megaco/H.248 are also viable (Kubher, 2004).

SIP offers a flexible and somewhat simpler implementation when building complex systems, whereas the more-established H.323 provides PSTN interoperability, a valued feature, and additional reliability. Standards bodies have been working on procedures to allow seamless internetworking between the two protocols. SIP was perceived as the emerging standard, with H.323 supposedly losing ground, but the older standard is still used by most service providers for control and signaling and for carrying public traffic (Shepard, 2005).

H.323. The H.323 standard from the ITU-T supports IP-based audio, video, and data communications. By complying with H.323, multimedia products and applications from different vendors can interoperate, allowing users to communicate without concern for software or system compatibility. The standard addresses call control, multimedia management, compression/decompression, and bandwidth management for point-to-point and multipoint conferences. H.323 also addresses interfaces between LANs and other networks (Khasnabish, 2003).

The H.323 standard includes other standards for compression and decompression of audio streams, ensuring that equipment from different vendors can communicate. As noted, the audio compression algorithms supported by H.323 are all proven ITU standards. Of these, G.723.1 operates at the lowest bit rate, while offering near toll-quality voice over IP networks. It operates at 6.3K bps and 5.3K bps, with selectable compression ratios of 20:1 and 24:1, respectively. G.723.1 also includes a silence compression feature, which compresses out pauses between spoken words. Silence compression can bring the effective rate down to 3.7K bps.

Session Internet Protocol (SIP). SIP is an IETF standard that uses text-based signaling to create IP calls. The protocol is independent of any applications and evolved from principles used with the Internet. The newer standard, which supports converged communications, including audio, video, and data, supports more of the next-generation services sought by service providers and enterprises, such as instant messaging; presence management; centralized call control; simplified moves, adds, and changes; voice-enabled browsing and e-commerce; and voice over packet. However, the commitment of large telcos to H.323, because of their massive ISDN infrastructure investment, has hampered the proliferation of SIP until such time as the telcos replace their legacy infrastructure. Figure 2 shows the schematic of a VoIP solution using SIP.

Security

One challenge and possible roadblock to implementing VoIP services is ensuring secure VoIP streams while maintaining QoS. A network security manager could easily combine traditional IP security techniques with payload and signaling encryption to secure VoIP calls. However, latency could be increased to the point where the quality of the average call would be unacceptable. Network security managers should expect security standards for VoIP to lag behind the technology, with lower security levels than time-division multiplexing (TDM) and PSTN for some time to come. Some of the major threats VoIP users face include the following:

• Phreaking. A form of hacking with the intent to steal or assume a VoIP user’s identity.
• Denial-of-Service Attacks. These attacks are malicious attempts to bombard the network with call-signaling messages to claim a large portion of available bandwidth, which then reduces the quality of service for VoIP, rendering Internet calls unserviceable.
• SPIT. Just as one can receive spam e-mails, VoIP spam can be sent in the form of voice mail.
• Viruses. The use of softphones, which are software products running on a PC or PDA while connected to the Internet, increases the opportunity to attract worms and viruses.
• Tampering. Attackers can add noise to a VoIP transmission to spoil the call quality.
• Man-in-the-Middle Attacks. Savvy hackers can intercept voice packets and even assume the identity of one of the calling parties. (Khasnabish, 2003)

Firewalls

For VoIP traffic to pass into a corporate network, the firewall must be opened to allow for the use of multiple ports, which are required for a single conversation. One solution, according to an article from Business Communications Review, is to open up VoIP ports dynamically, allowing traffic through only when needed, a process that can reduce the performance of the relatively unstable technology. As the use of VoIP grows, firewall providers will have to offer products that support dynamic opening of gaps in the firewall. Such products include dedicated hardware firewalls and encryption clients in VPNs. (Hersent, 2004) In addition, limiting the number of available applications on the operating system for an IP PBX will limit the number of openings for outside attacks.

Outlook

First-generation Internet telephony products introduced in 1995 were of such poor quality that it was hard to imagine that VoIP would ever become more than a hobbyist’s toy. In recent years, however, many of the issues that plagued users of first-generation Internet telephony products have been addressed by hardware and software vendors with the goal of facilitating the growth of commercial, carrier-class VoIP services. Improvements are still being made, mostly to the back-office systems that are required to support commercial service implementations. As a result of these improvements, VoIP has become a viable service that not even traditional telephone service providers can afford to ignore (Blosser, 2003).

The market for VoIP services will continue to grow. In 2006, equipment manufacturers shipped more VoIP ports than legacy telephone ports for the first time. While many businesses and consumers have made the switch over to VoIP, there are still an estimated 233 million people in the United States with a broadband Internet connection but do not sign up for VoIP. That represents a huge potential market that will be aggressively targeted by companies like Vonage as well as the communications carrier who provide that broadband service. Voice over IP is gaining a lot of attention in consumer market, but it is a legitimate technology in the corporate world as well. Companies are using VoIP as an inexpensive way to connect offices around the globe. According to information gathered by Frost & Sullivan, all of this potential is expected to turn VoIP into a $4.1 billion industry by 2010.

VoIP can also help these call centers expand their capabilities by allowing them to provide multi-channel support. As discussed earlier, VoIP is a flexible yet robust technology, and the convergence of voice and data traffic onto a single pipe allows businesses to offer services like videoconferencing and unified messaging. Companies can expand their support structure to include the telephone, chat, conferencing, instant messaging, and more. They can also use the full capabilities of a VoIP solution to set up a series of virtual area codes, making it appear that a business has a local presence in all of its operating markets.

Recommendations and Conclusion

VoIP technology has been deployed in many areas throughout the world, but it works best on managed IP-based networks, where performance can be closely monitored and fine-tuned by an enterprise or service provider. Companies that have a private intranet are already running or experimenting with integrated voice-data applications and IP telephony. The technology has progressed to the point that few reasons exist for companies to delay the phased implementation of VoIP solutions, especially for intra-company communications.

Traditional long-distance carriers, incumbent local service providers, competitive local exchange carriers (CLECs), cable companies, and national ISPs have come to recognize that IP telephony will continue to grow in popularity and have invested billions of dollars on new networks to position themselves for the provision of advanced services. Any company that is interested in implementing VoIP, whether they are skeptical or enthusiastic, should seriously and soberly weigh the initial investment and the cost of related organizational changes with the benefits and return on investment before jumping in or running away. A good place to start would be to offer VoIP as an adjunct to PBX service when its capacity begins to runs out instead of breaking the budget on an expensive PBX upgrade. Another option is to initiate a pilot leveraging a hosted PBX service provider to help the organization evaluate the technology, gauge performance, and establish internal training methodologies. Once the cost-savings opportunity has been identified and the research for the technology infrastructure carried out, then the capital investment in VoIP solutions can be made.

Finally, enterprises should understand the complete costs that are necessary for deploying a VoIP system. They include costs for upgrading to a packet-switched network, purchasing and upgrading equipment, implementing improved security measures, and managing and monitoring the new system. These costs may seem prohibitive in the beginning, but many enterprises will find long-term benefits from their investments. One benefit will be the available applications, which go beyond simple savings on long distance (although long-distance prices are beginning to climb again). Enterprises should see VoIP as a means to offer unified messaging and enterprise applications and to use IP softphones as network terminals. One drawback to using VoIP for more advanced applications, however, is to open up the corporate network to more security threats. These threats should be combated through the use of firewalls, increased server security, VPN encryption, and enhanced monitoring techniques.

References

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