Buyer’s Guide: Choosing a VoIP...

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Buyer’s Guide: Choosing a VoIP Vendor Uncover a comparison of the top VoIP vendors in the market

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Section 1: What is VoIP? p.2

Section 2: Do I need VoIP? p.9

Section 3: How to buy VoIP p.16

Section 4: Which should I buy? p.23

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In this e-guide: Voice over IP (VoIP) technology is quickly becoming a popular mode of voice communication, but how do you choose the right VoIP vendor? Start your search with the basics: understanding what it is and how it can help you.

VoIP technology can help boost productivity and provides a point of entry for businesses that want to upgrade their legacy telephony systems to more versatile IP communications services.

This buyer’s guide will walk you through how VoIP has evolved for the enterprise, the benefits it can provide your business, and criteria to consider when choosing a VoIP vendor.

Uncover a comparison of the top VoIP vendors in the market, followed by in-depth product overviews, to help you choose the right one.


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Section 1: What is VoIP?

The evolution of VoIP for the enterprise Voice over IP evolved out of a need to combine low-cost, Web-based data streams with closed-circuit TDM telephony.


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Analyzing VoIP basics reveals telephony's evolution

Jon Arnold, Principal, J Arnold & Associates

Voice over IP in the enterprise has clearly emerged as the successor to time-division multiplexing (TDM), offering businesses an overhaul of their telephony systems. In 2015, VoIP reached its 20-year milestone, so the technology has been with us for a long time. But as often happens in technology, the adoption process has been slow.

VoIP has been business-grade for many years. Its gradual adoption has reached critical mass, where most businesses now know about it and a growing number are using it. Before delving into the buying process, it's important to cover the VoIP basics, especially if you've never used it before.

Breaking down the VoIP basics Let's start by breaking down the acronym, which incorporates two distinct ideas. Voice over Internet Protocol represents the intersection of telephony and the Internet. Telephony is the primary medium for voice -- but not the only one -- and has long operated independently from other communication channels. This approach served businesses well, as there was no need to integrate it with anything else until the Internet came along.


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The Web has given rise to the Internet Protocol (IP), a set of standards that allows various forms of data to travel over a common network. Since legacy telephony -- TDM -- was not data, it initially remained separate, running on a dedicated voice network.

The common denominator for IP traffic is packetization, which transports digital packets across the Web in a highly efficient manner. The information flows at wire speed this way, with virtually no cost. Because the Web is a collection of interconnected networks, the operating principle is for packets to be routed over the most efficient path that consumes the least amount of network resources. Since many paths are possible, traffic rarely flows in a linear fashion, often making multiple hops en route to the final destination.

This approach is the polar opposite of TDM, which operates on the principle of a closed circuit between two parties. By having a dedicated connection, TDM calls are always clear, private and secure. But those circuits cannot be shared while in use, and compared to IP, TDM is a highly inefficient model for managing network resources. This is a key reason why the cost of legacy telephony is so high.

VoIP's pioneers recognized the opportunity and worked on bringing these worlds together. Once they could find a way to packetize voice, it could be added to the IP data stream and, from there, dramatic cost reductions in telephony service became possible. Early efforts were plagued by poor quality, but as the Web matured and broadband supplanted dial-up service, VoIP overcame those issues. Today, it is a bona fide replacement for legacy telephony.


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Enterprise VoIP basics and components When purchasing VoIP, businesses must understand two components are involved: the phone system and the service. They can be purchased together as a service, in which case there won't be any integration issues since the phones will already be IP-enabled. As explained above, VoIP travels on the data network, and the phones must be IP-based for the packets to convert into voice signals. Legacy-based phones are built around analog or digital technology, but will only work on the dedicated voice network -- not an IP network.

Although many businesses have sunk costs in legacy phone systems that are still going strong, they can still use VoIP -- so these businesses are not prevented from starting down this path now. To IP-enable these phones, you need an Analog Telephone Adapter (ATA) or comparable peripheral for each endpoint. ATAs aren't expensive and provide a great bridge to continue using legacy phones until it's time to replace them -- along with getting the cost savings from VoIP today.

For businesses that have no existing telephony system configured, going with a full VoIP service and IP phones makes sense. With phone vendors and service providers both transitioning from legacy to VoIP, greenfield opportunities have many options to consider.

A different scenario could be a trigger event where the existing phone system needs to be replaced. Whether a breakdown occurs or capacity


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issues simply reach a point of no return, a change has to be made. This will likely lead to a wholesale switch for both the phones and the service.

How enterprise VoIP is sold With legacy telephony, the two components -- phones and service -- were separate and not packaged as a system. Your incumbent carrier had a monopoly on the service, so there was no question about where to buy. Similarly, incumbent telephony vendors managed to keep customers captive with proprietary phone systems that were complex, inflexible and costly. In short, switching costs were high and choice was limited.

VoIP presents a broader playing field, mainly due to standardization. A core hallmark of the Web is using open systems that support interoperability. As a result, all IP phone systems will work with all VoIP services, so businesses can mix and match in ways they never could before.

Another twist with VoIP is that vendors can offer the service as well. The lines are blurring between carriers and vendors, as hosted VoIP is offered by a range of providers. Most telephony vendors offer their own turnkey VoIP services, hosted either in their own data centers or via third parties.

Businesses can choose to buy each component separately or as a service -- either from a vendor or a carrier. Each carrier has partnerships with IP phone vendors -- with a range of models and price points to choose from to make an all-in-one purchase. Beyond that, other channels cater to other types of buyers.


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For small businesses -- especially small and home offices -- VoIP can be purchased online and, in some cases, even provisioned and managed. At the higher end, businesses can buy via dealers or resellers who are more involved with evaluating services to assist with the decision-making and deployment.

Why businesses buy VoIP The prime driver for VoIP has always been cost savings, especially early on when legacy telephony was more expensive. Competition has driven prices down, so the savings are not as significant -- but still worthwhile. Many variables go into the total cost of telephony, but businesses can expect to reduce this expense in the range of 20% to 30%.

As the cost advantage shrinks -- and as IP communications mature -- businesses are starting to take a more strategic view of VoIP. A key benefit, aside from saving on telephony costs, is network convergence, where voice traffic is ported to the data network. With VoIP, telephony becomes a data stream, so there's no longer a need to maintain a separate network. This creates another form of cost savings by consolidating all media streams on to a single network.

Network convergence also creates an opportunity for IT to provide employees with a richer and more flexible set of communications applications. VoIP has a broader feature set than basic telephony, as it can integrate voice with other applications like click-to-call. This allows IT to


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switch from a costly, limited form of telephony to a richer alternative that does more than simply replicate TDM.

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Section 2: Do I need VoIP?

How buying VoIP benefits businesses Buying VoIP provides many benefits for businesses, including cost savings, easier migration to unified communications, and higher productivity.


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Three use cases that highlight the benefits of VoIP


Once a business realizes VoIP's potential, many questions may be raised -- not just about the technology, but also its business value. New technology usually implies risk. But in the case of VoIP, that concern should be nominal.

While many businesses are still learning about the benefits of VoIP, the technology is mature enough to replace legacy telephony. This consideration is important since the underlying time-division multiplexing (TDM) technology has served businesses so well for so long.

When understanding the need for VoIP, this history is relevant because the bar was set so high. TDM is expensive, but delivers outstanding quality and reliability. For IT departments, TDM does not pose any major network challenges, and for management, the business case has to be strong to consider replacing something that works so well.

Migration to VoIP usually requires a breakdown in an organization's telephony -- either a system failure or a capacity limit. In this scenario, keeping the phones running usually trumps other needs. But in most situations, three common VoIP use cases drive the migration.





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VoIP use case #1: Reduce telephony costs Curbing costs is the standard use case for VoIP, resulting in savings of approximately 20% to 30%. Enterprises usually switch to VoIP as a cost-savings measure for two reasons. First is a specific need to cut telephony costs, especially as free or near-free services like Skype become more prevalent in the enterprise. Second would be a broader mandate from management to reduce operational costs, in which case anything under IT's domain would be fair game.

Moving to VoIP will address these objectives right away, but longer term, the magnitude of savings will diminish. Businesses will realize immediate savings for the service itself, as the monthly line rate will be less than TDM, and many enhanced features incurring additional charges with standard telephony are included free with VoIP. Furthermore, with VoIP, domestic long-distance calls would essentially cost nothing and international calls will certainly be less.

Economically, this adds up to bottom line savings, but that must be contrasted with the overall cost of migration to VoIP. On the plus side, notable cost reductions exist with network convergence, where telephony traffic shifts from the dedicated voice network to the data network. Migration to VoIP also has the potential for hardware savings, as the cost of IP-based phone systems and handsets will be less than continuing with legacy systems.




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However, there may be hidden costs with network upgrades to support VoIP on the LAN, and possible penalties for ending existing contracts with telephony vendors. Enterprises upgrading from a legacy telephony system to a VoIP setup would need to expand their network capacity to carry voice traffic and possibly add SIP trunks.

Furthermore, new network management capabilities would be necessary to prioritize voice traffic and provide appropriate VoIP security protection. Finally, the organization would need VoIP-specific network infrastructure such as media gateways and session border controllers. Overall, lowering costs can be a solid use case for VoIP, but only if the switch is made with these factors in mind.

VoIP use case #2: Hit the refresh button Businesses cannot stay competitive with legacy technology, and VoIP can be a catalyst for change. On a functional level, legacy technology may be reaching its natural limits, forcing enterprises to replace their aging TDM systems. Typically, businesses have a telephony environment that works well, but only for its intended purpose. However, once employees adopt new technologies on their own, legacy telephony usage falls off.

Compared to other voice modes -- either Web-based or mobile -- TDM cannot support the way people communicate today. Mobile work styles are becoming the norm, and employees cannot be truly productive when tethered to their desk phones. While VoIP is still emerging on mobile devices,


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PC-based modes are common and can go a long way to supporting employees working away from their desks.

Furthermore, employees increasingly rely on other Web-based modes such as email, chat and video, and VoIP natively integrates with these, whereas TDM cannot. As the drop-off in desk phone use becomes more evident, the benefits of VoIP for business are clear.

A more strategic use case is the overall network environment. Unlike legacy telephony, VoIP is not a standalone application. Beyond supporting real-time voice, VoIP adds value by integrating with other data applications over the LAN. This takes VoIP to a higher level, where voice can seamlessly mix with video and text-based modes, not just for internal communication, but when dealing with customers as well.

VoIP also provides businesses with the opportunity to migrate their phone management to the cloud. Legacy telephony has been managed on premises, but hosted options based in the cloud are gaining favor for VoIP, which may lead to a broader type of technology refresh. Cloud-based services further expand VoIP's versatility.

Businesses may not see these possibilities until they take a deeper look at VoIP -- and when that happens, attention will shift to the network, raising new questions about how best to support all this. Rather than simply updating an existing telephony system, VoIP can lead a business to look at a broader technology refresh, especially around the investment needed to deliver these capabilities.


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VoIP use case #3: Improve productivity This use case builds on the previous one, but can also stand on its own to support a VoIP buying decision. Here, however, the focus is on employees rather than what's best for IT and the network.

For most businesses, productivity is subjective, so the use case is difficult to quantify. Metrics do exist for this, but they are not widely used outside the contact center. VoIP can make employees more productive in various ways.

Two prime examples would be visual voicemail and ad hoc conferencing. VoIP provides email alerts with MP3 audio files for voice messages, making it easier for employees to manage missed calls from anywhere. With today's dispersed workforce, ad hoc audio conferencing enables collaboration in ways that TDM could never support.

The real-time nature of voice makes collaboration more efficient when compared with near-real-time applications such as email or chat. When businesses make productivity a cornerstone of becoming more competitive, this VoIP use case will become self-evident for management.

Think short and long term VoIP isn't the only way to boost employee productivity, nor is it the only reason to upgrade the network, but the use cases can be very strong. This largely depends on whether the decision to switch is driven by IT or


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management. The overall takeaway is VoIP can provide short-term and long-term strategic benefits for the organization.

Thinking about VoIP strictly as a vehicle to save money is a valid driver, but a narrow one that undermines the bigger picture value proposition. Businesses had no reason to think this way with standard telephony, but they do with VoIP. Throughout the rest of this series, we will outline the factors to consider -- including the benefits of VoIP -- when making a purchasing decision.

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Section 3: How to buy VoIP

Criteria to consider when choosing a VoIP vendor Buying VoIP products requires careful consideration of a number of factors. VoIP vendors should be evaluated primarily by the following criteria: ease of use, definition of scope, existing network infrastructure and the possibility of future upgrades.


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What you need to consider before you buy VoIP


Selecting a VoIP service provider requires careful consideration for two distinct reasons. First, you need to determine from whom you'll be buying. In most cases, you will buy VoIP via a reseller or dealer -- the channel -- but in some cases, businesses buy VoIP direct from the vendor. Secondly, buying a VoIP phone system has more variables than a legacy phone system. In some cases, the need is for a full-fledged IP phone system, while others might just want basic IP phones.

Furthermore, IP phones are increasingly bundled with other things, such as the VoIP service, broadband connectivity and SIP trunking. A comprehensive set of buying criteria will depend on the needs of the organization, but the four common criteria that will apply to all VoIP phone system scenarios are: VoIP features, scope of the service, network requirements and plans for future expansion.

Ease employees into VoIP features and functions Considering features and functions should always be the starting point when you are ready to buy VoIP. To get full value, products should be compared with modern VoIP offerings rather than an organization's existing time



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division multiplexer (TDM) service. At minimum, VoIP must replicate legacy telephony, especially to ensure easy adoption among employees. The phones should have a familiar look and feel with the dial pad, preset buttons, handset, visual display and volume controls.

Beyond this, however, VoIP supports new features such as ad-hoc conferencing, call recording, Web browser, presence indicator, Bluetooth support and more. Since these phones connect to the data network, there will be different power configurations. Some models will be powered by plugging directly into the power supply, but most will use Power over Ethernet.

In terms of must-have VoIP features, the most important factor is ensuring the phones support the features being offered by the VoIP provider. Some phone vendors still maintain proprietary protocols, especially for SIP trunking, so choosing a vendor should not be done in isolation from the VoIP service. Another example is high-definition voice, which can be a great feature for the contact center or conferencing, but only if the endpoints can support it.

Finally, the range of VoIP features varies by price point, and IP phone vendors offer a range of models to address different use cases. While most employees can manage with a standard phone model, executives might want higher-end phones. Contact center agents need models with good headset support. Warehouse employees need more rugged and/or cordless phones. And meeting rooms require conferencing phones. When developing a request for proposal, identifying the various needs will be a key area for vendors to tailor their offer.



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Grasping the scope of VoIP service Unlike TDM, VoIP is not restricted to the desk phone and can take several configurations. This is a key area where telephony consultants add value, as businesses that are new to VoIP may not understand the range of possibilities. Legacy telephony was straightforward since the service ran on a dedicated voice network, but VoIP supports connections to a variety of technologies through the Internet.

Organizations that purchase VoIP will at least need a standalone phone system, such as when the business has adequate network support for VoIP and does not need SIP trunking. This would be a bring your own broadband model, where the vendor provides a phone system that IT manages.

However, even this scenario will require additional network elements to support VoIP, such as media gateways and session border controllers. Furthermore, businesses may elect to keep some legacy phones and VoIP-enable them; in that case, Analog Telephone Adapters are needed.

Another consideration is maintenance and professional services, where the range of offerings is quite broad depending on the support the business needs. Prime examples include system design, installation, system configuration and training -- for both IT and end users.

Additionally, there is the option of either cloud- or on-premises-based, and this can have a major impact on the specifications. These are just a few



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examples of what scoping out the service entails. And for first-time VoIP buyers, the importance of doing this up front cannot be understated.

VoIP could require network overhaul When considering the overall operational requirements for VoIP, buying a phone system is only half the equation. Moving to VoIP will trigger a series of network-related questions, as businesses can cut over completely or in phases. This decision can have major implications for the project's scope, as VoIP marks the beginning of a transition away from having a separate network infrastructure to support telephony.

Ideally, the data network is able to absorb VoIP traffic, in which case only the new phone system needs to be acquired. More likely, however, the LAN will need some form of upgrade or expansion.

While VoIP is not a bandwidth-intensive application, adding a lot of traffic from the start could impact performance. More importantly, data networks are not designed to support real-time modes such as voice, and this is where IT faces the biggest challenge. Aside from possibly needing to upgrade cabling or capacity, new network management tools and hardware may be needed to prioritize voice traffic and ensure quality of service (QoS).

Another variation would be to route voice over a separate data network. Some businesses do this to achieve a higher level of data security, as VoIP traffic can be an easy target for cyberattacks. While this may seem to negate all the benefits of taking voice off the dedicated TDM network,


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businesses do get all the benefits that come with VoIP, but in a more secure manner.

Beyond this, there are additional network-related considerations around SIP trunking and Wi-Fi. Both can enhance the value of VoIP for the business but have implications on the network and infrastructure required to support the phone system. As such, when developing buying specifications, the network needs must be taken into account.

Plans for future expansion with VoIP Legacy phone systems were bought to do one thing, so there is little need to think about their future. TDM innovation has been stagnant for decades, but VoIP is evolving. Today, businesses do not buy VoIP phone systems with a 15-year horizon. And businesses should keep in mind that the role of voice communication is not defined solely by VoIP.

In the broader context of mobile broadband, WebRTC and cloud communications, businesses need to define their VoIP objectives beyond just buying a phone system. The business value of VoIP is based as much on the utility of desk phones as the ability to integrate VoIP with other applications and communication modes. To address this, telephony vendors have higher-end models built to support multimedia requirements, and some offer docking stations to support tablets and smartphones for office use.


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On the network front, most businesses still prefer to be based on premises or have a mix of on-premises and cloud-based systems for their VoIP environment. Only a minority is currently ready for a full cloud migration.

And when planning for VoIP, the safest course is to partner with a vendor that can support a hybrid service. Vendor capabilities vary widely, so businesses need to determine which services best align with their long-term needs.

Making decisions when you decide to buy VoIP can be simple or complex, but in most cases, there will be several considerations beyond the basic phone system. VoIP features bring exciting opportunities to any business, but the range of deployment scenarios can be daunting. For this reason, choosing the right vendor is critical, especially since apples-to-apples comparisons cannot be made for every attribute.

If you're replacing TDM with VoIP and integrating VoIP features with a broader communications landscape, you will have more success if you develop a clear set of needs.

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Section 4: Which should I buy?

Choosing the right VoIP product for you Take a look at our comparison of the leading VoIP vendors on the market before you make your purchasing decision. Uncover product overviews of the leading VoIP vendors. Many of these products can accommodate a range of business sizes and enhance the way companies work.


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Comparing the leading VoIP vendors and their products

Chris Partsenidis, Founder, Firewall.cx

Today's IP PBX market offers a range of services that should satisfy almost any business requirement. But selecting the right IP PBX for your business can be a complicated process, mainly because of the various IP PBXs, features and services offered.

You should consider four major criteria when purchasing an IP PBX product: VoIP features and functions, an end-to-end service vendor, redundancy capabilities, and expanding user and device support. Let's take a look at some leading VoIP vendors and examine how their products fit into these categories.

Important VoIP features and functions SIP trunking is a feature that should be present on any IP PBX today. Some vendor models, such as the Mitel 3300 series, can support up to 2,200 SIP trunks. NEC's Univerge can support up to 4,000 SIP trunks, while Cisco's CUCM can exceed these figures with the appropriate VoIP infrastructure.

Toshiba's IPedge can support up to 440 SIP trunks. Panasonic's KX-TDE supports up to 64 SIP trunks. ShoreTel and Adtran do not specify their


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maximum support for SIP trunks, but I'm sure they would be able to support an SMB.

Businesses with call-center services might want to consider systems with a high number of SIP trunks, as they usually require connections to multiple providers, both local and international.

Do early research on IP phones Phone design and functionality play a critical role in the buying decision. VoIP vendors such as NEC, Panasonic, Mitel and Toshiba let customers select IP phones or digital phones. This is a major advantage, especially for businesses upgrading from a PBX to an IP PBX, as it might be possible to re-use their existing digital or analog phones with the new IP PBX.

On the other hand, vendors such as Cisco, Adtran and ShoreTel offer only IP phones as part of their portfolio, though customers can also connect analog phones to the IP PBX using analog telephone adapters (ATAs).

In terms of variety, Cisco and Mitel are the only two vendors that offer more than 25 different IP phone models -- from simple single-line IP phones to multi-touch color LCDs. Cisco's top IP phone models also offer video capabilities.

NEC's UT880 IP phones are also impressive, offering full unified communications (UC) functionality geared for executive users. ShoreTel


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offers more than 15 IP phone models with similar functionality as other vendors.

Because of the large variety of IP phones, businesses need to examine the phones offered by each vendor before proceeding with the purchase of their IP PBX. Also note that some IP phone models are only compatible with the IP PBX service from the same vendor. So, if you want to mix and match different vendors, ensure the phones are compatible with the service.

Some warnings about remote user support User mobility is a growing trend and supported by all vendors, but some caveats need to be examined. All VoIP vendors offer a proprietary mobile application that is available for the Windows platform -- for mobile laptop users -- and iPhone and Android users. Yet, in almost all cases, this setup requires a virtual private network (VPN) so remote users can connect to their headquarters before logging on to the IP PBX system.

ShoreTel provides mobility support via its dedicated mobility router appliances, though it requires purchasing additional hardware and licenses. Cisco can integrate VPN support on its voice gateway, which requires an additional security license and offers a mobile VPN-less option based on its Expressway servers -- all part of the Business Edition platform. The VPN-less approach requires additional licensing and hardware infrastructure in the form of a firewall and router.


http://searchenterprisewan.techtarget.com/definition/virtual-private-network

http://searchnetworking.techtarget.com/definition/gateway

http://searchsecurity.techtarget.com/definition/firewall

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Mitel's 3300 CXi II controller offers a wide area network (WAN) port for direct Internet connectivity, supporting network address translation (NAT) and firewall services, but lacks the ability to support VPN. Adtran does not offer support for Internet connectivity, firewall and VPN services with its NetVanta 7060 platform.

Voicemail port support Voicemail is considered a standard feature of any IP PBX. While all systems support voicemail features, you should ensure your system can support enough mailboxes to meet your organization's needs. You should also ensure it offers enough voicemail ports, which identify how many simultaneous users can log into the system and listen to their messages.

NEC's SV8100 supports more than 500 mailboxes but offers only up to 16 maximum voicemail ports -- NEC calls them voice response channels. Mitel's 3300 CX II can support up to 750 mailboxes and 16 voicemail ports. Some vendors do not clearly state how many voicemail ports their systems support.

VoIP vendors with and without end-to-end services Only two vendors -- Cisco and Adtran -- currently provide an end-to-end solution.


http://searchenterprisewan.techtarget.com/definition/WAN

http://searchenterprisewan.techtarget.com/definition/Network-Address-Translation

http://www.adtran.com/web/page/portal/Adtran/product/1700706G1/15

http://whatis.techtarget.com/definition/voicemail

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Cisco offers a range of products for IP PBX systems, IP phones, voice gateways and network infrastructure. Cisco's network switches include Power over Ethernet (PoE) options, quality of service (QoS), Layer 3 switching for interVLAN routing and network segmentation for added security.

Adtran is the only vendor that comes close to an end-to-end solution like Cisco. The NetVanta 7060 provides a built-in 24-port Layer-2 10/100 Gigabit Ethernet (GbE) PoE switch. However, it has no 1 GbE support and no support for Internet connectivity. In addition, the NetVanta line includes IP phones, Ethernet switches, routers and wireless access points.

Some VoIP vendors provide separate or built-in products worth noting, though these are not full end-to-end services. Mitel's 3300 CXi II controller integrates a 16-port Layer-2 10/100 GbE PoE switch and support for firewall and Internet connectivity. ShoreTel sells routers and VPN concentrator devices, but does not offer any network switches.

Redundancy combats network failures Redundancy is an essential ingredient to any business that cannot afford downtime and runs mission-critical services via its communication infrastructure. IP PBXs that offer an all-in-one package seem to attract businesses because of their lower costs -- but they usually fail the redundancy test.



http://searchunifiedcommunications.techtarget.com/definition/QoS-Quality-of-Service

http://searchnetworking.techtarget.com/tip/Layer-3-switches-explained

http://searchnetworking.techtarget.com/tip/Layer-3-switches-explained

http://searchnetworking.techtarget.com/answer/What-is-interVLAN-routing

http://searchnetworking.techtarget.com/definition/segmentation-and-reassembly

http://www.telewire-inc.com/productcatalog/mitel/pdf/Mitel3300Controllers-DataSheet.pdf

http://searchnetworking.techtarget.com/answer/How-does-the-VPN-concentrator-work

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Communication failure does not necessarily occur only at the IP PBX level. WAN link failure is a common example, leaving remote sites without any telephony systems and intra-site communication.

Cisco's VoIP infrastructure allows multi-level redundancy at a higher cost. Cisco CallManager -- the main IP PBX product -- supports clustering, allowing multiple instances of CallManager to work together in a failover configuration.

Cisco's voice gateways (ISR routers) connect the IP telephony infrastructure with telecommunication providers, but can also serve as a second backup device should the CallManager cluster fail. This is called survivable remote site telephony (SRST). When such a failure occurs, IP phones will automatically register to the voice gateway for basic phone functionality until the CallManager server or cluster is restored. During the failure of CallManager server or WAN link connecting the sites, remote site IP phones registered to the main office's CallManager servers will fall back to their local voice gateway, which will provide SRST services.

ShoreTel also offers products with redundancy capabilities. IP phones register with their local voice switch. When this voice switch fails, IP phones automatically register with a redundant voice switch that has been installed elsewhere in the network -- locally or at the headquarters. This transition happens automatically, minimizing downtime across the enterprise.

Mitel's 3300 controllers offer similar redundancy when deployed as part of the Mitel MiVoice Business service. Businesses can deploy a centralized IP


http://www.cisco.com/c/en/us/products/unified-communications/unified-communications-manager-callmanager/index.html

http://searchstorage.techtarget.com/definition/failover

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http://www.mitel.com/products/mitel-mivoice/mivoice-business

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PBX system similar to Cisco's CallManager while also providing SRST services to remote sites in the event of a WAN link failure.

The Toshiba IPedge EP server can be configured in a cluster mode, where each IPedge server is an individual node connected with the rest of the cluster, via the WAN or LAN. One server is configured as the primary node and centralized administration is performed via the primary node's Enterprise Manager. IP phones are configured to fall back to a secondary IPedge server when the primary fails.

Other VoIP vendors offer decentralized call processing, and therefore focus more on local system redundancy -- in case there is more than one IP PBX system installed. If redundancy is important to your business, be sure to identify which redundancy options you need.

Accommodating for business growth Expanding your IP PBX system to support more users and devices as your business grows is important, as it helps curb costs and allows businesses to pay as they grow.

Mitel can support up to 999 controllers clustered together, so businesses simply add controllers and licenses to support the required amount of users. The milestone here for an additional hardware controller depends on your PBX model, starting with 150 users for the smallest Mitel Controller, the 3300 CX II.



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NEC works along the same lines with its smallest IP PBX model -- the SV8100 -- supporting up to five IP PBX units stacked and around 90 to 100 phones per unit. The total number of phones supported per unit depends on the type of phones used.

Panasonic offers similar user expandability, but only with its KX-TDE600 model, which supports the base unit -- the main PBX -- plus up to three expansion shelves. Unlike other vendors, with Panasonic you simply purchase an expansion shelf and applicable licenses. The maximum number of phones supported by each expansion shelf depends on the type of phones.

ShoreTel supports user expansion by simply purchasing another voice switch, but does not specify how many voice switches can be added to a single site. I believe the number of voice switches depends on the model and business infrastructure.

Cisco's CallManager BE6000 can handle up to 1,200 devices with its medium-density servers, while Cisco's high-density servers push this limit to 2,500. To reach these limits, businesses must buy additional BE6000 servers, but the milestone for each server depends on the system's configuration and services offered. Licenses will also need to be purchased.

Toshiba offers a maximum of 128 IPedge servers in a network, allowing businesses to expand their user support to thousands. The exact figure depends on the overall system configuration, however the IPedge EP model offers support of up to 40 users per server, whereas the IPedge EM -- the


http://www.necam.com/docs/?id=8c037fb7-7318-4ea8-9814-cedace3ba6a7

http://business.panasonic.co.uk/communication-solutions/sites/default/nfsfiles/communication-solutions/files/specsheet_uploads/TDE600.pdf

http://business.panasonic.co.uk/communication-solutions/sites/default/nfsfiles/communication-solutions/files/specsheet_uploads/TDE600.pdf

http://www.cisco.com/c/en/us/products/collateral/unified-communications/business-edition-6000/data_sheet_c78-717454.html

http://www.telecom.toshiba.com/mobile/products/ipedge/specifications/

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largest available model -- supports up to 1,000 users per server. In addition to the hardware, additional user licenses will be required.

Find the service that fits your business All of these products have pros and cons that you should compare with your organization's needs when purchasing an IP PBX system.

For large enterprises or expanding mid-size businesses looking at a complete IP PBX service with accompanying LAN infrastructure, the comprehensive services offered by Cisco, Toshiba and Mitel are probably your best bet. A small company with a basic LAN infrastructure that does not need VLAN segmentation and QoS might want a more hybrid service offered by Panasonic, NEC or Adtran.

Businesses must keep in mind that when moving to an IP-based PBX that involves IP phones, it is imperative their network infrastructure is VoIP-ready to ensure a smooth transition to the world of VoIP.

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http://searchunifiedcommunications.techtarget.com/tip/Five-techniques-for-getting-a-robust-VoIP-infrastructure

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Adtran VoIP offerings: NetVanta 7100/7060 product overview


The NetVanta series is the main Adtran VoIP offering, available in two models: the NetVanta 7060 IP PBX and NetVanta 7100 Unified Communications IP PBX.

The NetVanta 7060 is geared toward small and medium-sized businesses (SMBs), offering support for up to 100 IP phones. The all-in-one box solution is ideal for companies that are looking for a simple and complete VoIP system that won't break the budget.

The NetVanta 7060 includes support for SIP trunking, though the maximum number of SIP trunks is not specified. The Adtran VoIP product also features a built-in multilevel auto attendant supporting complex menus, and voicemail support with eight voicemail ports, which means up to eight users can simultaneously access their voicemail.

The NetVanta 7060 also offers support for two FXS ports, allowing two analog telephones to connect to the system, two FXO trunk ports (for PSTN lines) and two module slots that can be populated with a variety of Voice Interface Modules (VIMs). These include a four-port analog FXO trunk module, a four-port FXS station module (for phones), a T1/PRI trunk module, and a number of other VIM cards.



http://www.adtran.com/pub/Library/Data_Sheets/Default_Public/NetVanta_7060_Data_Sheet.pdf

http://searchunifiedcommunications.techtarget.com/definition/IP-telephony


http://whatis.techtarget.com/definition/auto-attendant-automated-attendant

http://searchunifiedcommunications.techtarget.com/answer/What-is-FXO-and-FXS

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http://www.adtran.com/web/page/portal/Adtran/group/16

http://www.adtran.com/web/page/portal/Adtran/group/16

http://searchenterprisewan.techtarget.com/definition/T1

http://searchnetworking.techtarget.com/definition/Primary-Rate-Interface

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Additionally, two small form-factor pluggable and two Gigabit Ethernet ports feature uplink capabilities.

The NetVanta 7100 offers similar features as the 7060 model, but the 7100 is a more complete unified communications product. It includes an IP router, stateful firewall capabilities and more.

One of the areas where the Adtran VoIP series falls short is the built-in 24-port power over Ethernet 10/100 Mbps switch. Ten years ago, this would not have been a big problem; but today's networks hardly use 10/100 Mbps switches. And when you consider that most of Adtran's IP phones have a built-in Gigabit Ethernet switch port, you realize the built-in switch is pretty much useless for customers who want to use Gigabit Ethernet.

However, unlike the NetVanta 7060, the 7100 offers a built-in wireless network access controller used to configure and control NetVanta wireless access points. This much-welcomed feature truly confirms this is an all-in-one product for SMBs.

NetVanta series Adtran VoIP products can be managed via the built-in Web interface from any workstation that has direct access to a NetVanta IP PBX.

Integration with Microsoft Outlook and other applications is possible via the Adtran Business Communications System, a software-based UC solution with unified messaging, fax and email.


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http://searchnetworking.techtarget.com/definition/Gigabit-Ethernet

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http://www.adtran.com/pub/Library/Data_Sheets/International_/NetVanta_7100_Data_Sheet_International.pdf

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http://www.adtran.com/pub/Library/Data_Sheets/Default_Public/61951050SAG1-8_NV_Business_Comm_Sys.pdf

http://searchmobilecomputing.techtarget.com/definition/unified-messaging

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Adtran IP phones The Adtran VoIP line also includes a wide range of IP phones and conference stations that should satisfy most users. Most IP phones offer a dual-port Gigabit Ethernet switch port, allowing the simultaneous use of a computer and IP phone through one Gigabit Ethernet port while using 802.1q trunking.

The high-end IP phones offer 12- or 16-line support with a 3.5-inch or 4.3-inch color multi-touch display, and the ability to add a camera to support video conferencing. Lower-end IP phones offer a decent LCD display with two- to six-line support and Fast Ethernet or Gigabit Ethernet support.

Adtran also offers an IP softphone available only on the Windows platform, though users can install almost any SIP-based IP phone on their mobile device.

Adtran VoIP products are available through its official distribution and partner channels. Customers seeking an Adtran solution can also contact Adtran's sales team directly.

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Cisco UCM voice over IP system: Product overview


Cisco's Unified Communications Manager (Cisco UCM) is an IP-based communications system that supports voice, conferencing, video, data, instant messaging and mobility services. With Cisco UCM, businesses can replace their legacy PBX telephone system with a modern, secure VoIP system that can scale beyond a legacy PBX and provide unified communications services to multiple locations.

Because of the modular design of Cisco UCM, it is suitable for virtually any size company. It uses a server-clustering design, which allows each cluster to reach 30,000 users. Organizations can deploy multiple clusters, significantly increasing this number to well over 100,000 users. Hardware requirements for a Cisco UCM (CUCM) deployment will depend on the organization's size and which add-ons are selected.

CUCM is available as a standalone product or part of the Cisco Collaboration Solutions -- Cisco Business Edition 6000 (BE6K) and 7000 (BE7K). The BE7K edition is a family of end-to-end collaboration services designed for fast-growing enterprises, while the BE6K is targeted toward enterprises ranging from 25 to 1,000 users, and provides the necessary applications to support business growth. Both BE7K and BE6K are offered on a virtualized platform running under the VMware hypervisor.



http://searchunifiedcommunications.techtarget.com/definition/private-branch-exchange




http://docwiki.cisco.com/wiki/Virtualization_for_Cisco_Unified_Communications_Manager_%28CUCM%29

http://www.cisco.com/c/en/us/products/collateral/unified-communications/7800-series-media-convergence-servers/prod_brochure0900aecd8062a4f9.html

http://www.cisco.com/c/en/us/products/collateral/unified-communications/business-edition-6000/data_sheet_c78-717454.html

http://www.cisco.com/c/en/us/products/collateral/unified-communications/business-edition-7000/data-sheet-c78-730649.html

http://www.cisco.com/c/en/us/products/collateral/unified-communications/business-edition-7000/data-sheet-c78-730649.html

http://searchvmware.techtarget.com/definition/VMware

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Cisco also offers the much smaller Cisco Call Manager Express (Cisco UCME) IP PBX service, which is targeted for small and medium businesses. Cisco UCME runs directly on Cisco's ISR routers and, depending on the router model, can support up to 450 users. More information on Cisco UCME can be found on Cisco's website. At the core of the system is the CUCM software that runs on either a dedicated hardware server or in a virtualized VMware environment. A standard Cisco UCM setup consists of the following minimum equipment:

1. Hardware or virtual server running Cisco UCM 2. A voice gateway (ISR Cisco Router), responsible for connecting Cisco

UCM with the telecommunications provider (PSTN/ISDN, SIP) 3. IP phones for the company's users

In addition to these, customers can select more servers or services to expand functionalities. These additional services are part of Cisco's Unified Collaboration portfolio. Cisco offers auto-attendant services through Unity Express (a module inside a voice gateway), Unity Server, or the Cisco Customer Response Solutions server.

CUCM provides more than just the basic functionality of an IP phone. With the use of IP Communicator or the newer Cisco Jabber, which requires Cisco Presence Server, users can extend their IP phone to their desktop, allowing them more comprehensive mobility and connectivity.

Users can also access the Web interface of Cisco UCM and change specific settings on their IP phone, insert speed dials and adjust other settings.



http://searchconsumerization.techtarget.com/definition/Cisco-Integrated-Service-Routers-Generation-2-ISR-G2

http://www.cisco.com/c/en/us/products/collateral/unified-communications/unified-communications-manager-express/datasheet-c78-732585.html

http://searchnetworking.techtarget.com/definition/broadband-voice-gateway


http://searchenterprisewan.techtarget.com/definition/ISDN

http://searchunifiedcommunications.techtarget.com/definition/Session-Initiation-Protocol



http://www.cisco.com/c/en/us/products/collateral/unified-communications/unified-communications-manager-express/product_data_sheet0900aecd806aefe4.html

http://www.cisco.com/c/en/us/products/collateral/unified-communications/jabber-windows/data_sheet_c78-704195.html

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Generally, Cisco UCM has been built to provide the end user with a range of advanced functions, such as Web/video conferencing, instant messaging and other capabilities.

The Cisco BE6K is the most popular edition and combines the following packages into a single appliance (rack server) running under VMware ESXi:

• Cisco Communications Manager • Unity Connection • Unified Presence or Attendant Console • Contact Center or VCS (Expressway) • Provisioning Manager

Cisco's flagship product licensing model has changed several times in the past decade due to the increase of services being offered. Licenses are installed on the Cisco Prime License Manager, which monitors Cisco UCM's license requirements and reports back license compliance or noncompliance. The current licensing model depends heavily on the type of VoIP and collaboration services companies require from Cisco UCM -- this is also reflected partially on the type of IP phone models used. For example, more advanced -- and expensive -- IP phone models have features such as video conferencing and touchscreen capabilities, and therefore require more advanced licenses.

Cisco's website outlines available license types and the functionality each license type covers.


http://searchvmware.techtarget.com/definition/VMware-ESXi

http://www.cisco.com/c/en/us/products/collateral/unified-communications/expressway-series/datasheet-c78-730478.html

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmfeat/CUCM_BK_F3AC1C0F_00_cucm-features-services-guide-100/CUCM_BK_F3AC1C0F_00_cucm-features-services-guide-100_chapter_0100101.html#CUCM_RF_LC43D736_00

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Cisco UCM has a variable pricing structure, depending on the features and functionality chosen. While CUCM can be purchased directly from Cisco's official distribution channels, it is highly advisable that customers contact Cisco directly for the most accurate information. Alternatively, customers can locate a Cisco partner using the Cisco Partner Locator to help guide them through the purchase and installation process.

When installed, Cisco UCM remains fully operational in demo mode for 60 days. During that period, Cisco Prime License Manager must be installed, along with all necessary licenses.

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KX-TDE Series Panasonic VoIP system: Product overview


Panasonic entered the IP PBX market with its popular KX-TDE series PBXs and KX-NS1000 model. The KX-TDE series has three different models: the KX-TDE100, KX-TDE200 and KX-TDE600, each targeted toward different company sizes and connectivity needs. The KX-NS1000 is an IP PBX solution that offers extensive unified communications (UC) services to its users. A list of Panasonic's IP PBXs is available on its website.

The KX-TDE100 and KX-TDE200 models offer similar capabilities. Both models can support up to 256 extensions and 64 SIP trunks, but the models differ in the number of extension types they support, such as SIP, analog or digital phones.

The KX-TDE100 is housed in a cabinet with six free slots, whereas the KX-TDE200's housing provides 11 free slots. Customers can buy additional cards, allowing them to connect the PBX with analog phones, Panasonic digital phones, telecommunications providers (ISDN/PRI/PSTN) and more.

The KX-TDE600 model is the largest PBX currently offered in the Panasonic VoIP lineup. The main unit offers 10 free slots but is able to accept up to three additional expansion shelves -- each shelf supporting up to 11 free




http://www.panasonic.com/my/business/communications/pbx/ip-pbx.html

http://www.panasonic.com/my/business/communications/pbx/ip-pbx/kx-tde100.specs.html






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slots -- for a total of 43 slots. The KX-TDE600 can support up to 1,152 extensions and 640 trunks, of which 64 can be SIP trunks.

Different configurations provide users with flexibility The KX-TDE series is offered in a custom-made cabinet with proprietary cards, unlike other vendor solutions that are PC server-based. This design lets Panasonic keep a modular approach, allowing customers to expand their IP PBX by using special cards that are inserted into free slots inside the main cabinet, which further reduces costs.

With the KX-TDE600, customers buy the main cabinet -- or base unit -- and depending on their infrastructure and needs, they can purchase up to three expansion shelves, which are physical cabinets that stack on top of the base unit and provide the necessary expandability. Expansion shelves are only available on the KX-TDE600 model.

TDE systems can also network together, allowing businesses to link up to 16 sites seamlessly without additional hardware, such as proxies or voice gateways.

All KX-TDE models natively support IP and SIP phones, allowing businesses to register Panasonic IP phones or even third-party SIP phones, called SIP extensions. However, Panasonic VoIP phones are not compatible with other SIP-based IP PBXs. Depending on the number of SIP extensions, businesses


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should ensure the appropriate number of licenses and hardware digital signal processors are purchased and installed into the IP PBX.

The KX-TDE models also support SIP trunking, which allows businesses to connect directly to SIP providers and route incoming/outgoing phone calls via the Internet or through a dedicated leased line to the SIP provider.

The Panasonic VoIP system can also record voice messages 24 hours a day. Businesses looking for advanced voicemail and auto attendant features will need to consider the KX-TVA series voice processing system.

Setup and administration of the Panasonic KX-TDE series is performed by using Panasonic's Maintenance Console, also known as PBX Unified PC Maintenance Console, which is installed on a PC that is directly attached to the PBX. However, the Maintenance Console can be installed as a Web server on a client PC directly attached to the PBX, which allows users to configure the PBX via a Web browser.

Exploring some unique user applications The Panasonic VoIP suite works with some interesting software to expand the functionality and services that businesses can offer to their users. The Outlook Add-in USB software is a plug-in that allows users to use many of Outlook's email and organization features to set the behavior of their extension -- a few examples are scheduling automatic call forwarding via Outlook Calendar and making calls directly from Outlook contacts list.



ftp://ftp.panasonic.com/businesstelephone/brochure/kx-tva50_kx-tva200_brochure.pdf

http://whatis.techtarget.com/definition/Web-server

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Another useful application is Panasonic's Phone Assistant software, which allows users to use their workstation desktop to view call-related information, such as caller ID, and perform simple point-and-click actions, such as answering calls or viewing colleague availability.

Phone Assistant Status allows business managers and supervisors to use real-time monitoring of all member telephony communication. With the help of Phone Assistant Status, managers can monitor up to 128 users while viewing their presence and availability.

Panasonic offers all PBX models and software through its authorized dealer network.

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Mitel VoIP 3300 series controllers: Product overview


The most popular offering in the Mitel VoIP lineup is the Mitel 3300 series of controllers, initially introduced to the global market in 2001. Since then, Mitel has continuously revised and expanded its functionality by introducing new features to ensure it does not fall behind in the ever-expanding IP PBX market.

The Mitel 3300 controllers are IP PBXs that combine full IP services -- such as SIP trunking, IP phone support, IP network and application integration -- as well as legacy connection support, including analog and ISDN/PRI. The controllers satisfy pretty much any type of environment and connectivity requirements.

Mitel currently offers five models: the 3300 CX II, 3300 CXi II, 3300 MXe III Standard, 3300 MXe III Expanded and the 3300 AX. All Mitel VoIP models offer clustering capabilities, allowing up to 999 controllers in a cluster by utilizing Mitel's synchronization technology.


http://searchunifiedcommunications.techtarget.com/tip/IP-PBX-Ten-reasons-to-switch

http://www.mitel.com/sites/default/files/MiVoice%203300%20Controllers.pdf

http://searchunifiedcommunications.techtarget.com/answer/SIP-trunking-vs-VoIP-Whats-the-difference




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Mitel 3300 CX II and CXi II The Mitel 3300 CX II and 3300 CXi II offer similar capabilities, supporting up to 150 phone devices depending on the hardware configuration, 2,000 SIP trunks, 16 voicemail ports and a generous 750 mailboxes. They also support up to eight 5550 IP consoles, which is a PC-based attendant console and administration application.

Both Mitel VoIP models provide a maximum of two embedded T1/E1 digital trunk modules, two embedded BRI modules, and a maximum of eight embedded BRI interfaces, making them adequate to cover most business communication environments. For organizations relying heavily on SIP, the 3300 CX II and CXi II can support up to 64 G.729a compression channels.

The 3300 CXi II integrates a network switch (layer 2), router and firewall, making it ideal for deployments where there is no existing switch or router infrastructure. The Ethernet-WAN port provides all router and firewall functionality required to connect the business to the Internet -- a suitable configuration for small offices.

Integrated voicemail and auto-attendant functionalities are supported by both the 3300 CX II and CXi II models. Advanced auto-attendant functionality is available through the Mitel contact center solution.


http://edocs.mitel.com/UG/EN/33_5550IP_R4.0_QRG_EN.pdf

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Mitel 3300 MXe III Standard and Expanded The Mitel 3300 MXe III Standard and Expanded models offer increased capacity and expandability, making them ideal for continuously expanding small and medium-sized businesses (SMBs) and large enterprises.

The Mitel 3300 MXe III Standard supports a maximum of 350 phone devices, of which up to 300 can be IP/digital phones, making the system an ideal candidate for SMBs. Alternatively, all 350 devices can be analog phones. An increased capacity of 16 5550 IP consoles are supported by the 3300 MXe III Standard model.

The 3300 MXe III Expanded supports a maximum of 1,500 phone devices, of which up to 1,400 can be IP/digital. Alternatively, all 1,500 can be analog phones. The exact figures depend on the type of IP/digital phones. Up to 24 5500 IP consoles are supported by the 3300 MXe III Expanded model.

In order for the 3300 MXe III Standard and Expanded systems to reach their maximum capacity of analog/digital phones, the Mitel SX-200 Bay Cabinet must be used. The SX-200 provides the necessary interfaces for the analog/digital phones, and in turn connects to the Mitel 3300 MXe Controller. Multiple SX-200 Bay Cabinets can be used, depending on the system's configuration and business requirements.

Both the Standard and Expanded models have 20 voicemail ports that can be expanded to a maximum of 30, if needed. With a maximum of only 750


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mailboxes, companies might be left a bit short if considering the 3300 MXe III Expanded model, mainly because it supports up to 1,500 phones.

Examining the connectivity side, the 3300 MXe III Standard supports a maximum of three embedded T1/E1 digital trunk modules, a maximum of three embedded BRI modules, 12 embedded BRI interfaces and 2,000 SIP trunks. The 3300 MXe III Expanded marginally extends connectivity options by supporting a maximum of four embedded T1/E1 digital trunk modules, three embedded BRI modules, 12 embedded BRI interfaces and 2,000 SIP trunks.

The Mitel 3300 MXe III Standard and Expanded models can support up to 128 and 192 G.729a compression channels, respectively, saving valuable bandwidth during SIP calls. These numbers also depend on the configuration of DSP modules installed on the systems.

Mitel 3300 AX The Mitel 3300 AX model is ideal for high-density analog requirements, making it suitable for the hospitality industry. It supports up to a maximum of 400 phone devices, of which 125 can be IP/digital phones and 288 can be analog phones, while businesses can connect up to eight 5550 IP consoles to the system.

The 3300 AX supports 750 mailboxes with 20 standard voicemail ports that cannot be expanded. The 3300 AX supports one embedded T1/E1 digital


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trunk module, one embedded BRI module, up to four embedded BRI interfaces, 2,000 SIP trunks and 128 G.729a compression channels.

All Mitel VoIP products covered here are easily managed via a Web browser.

Mitel IP phones Mitel offers an impressive range of IP phones and softphones, designed to cover every working environment.

The executive IP phones offer a high-resolution color touch display, 48 programmable multifunction keys, superior sound quality and a number of useful built-in features, such as support for cordless (DECT) and Bluetooth accessories, and support for various Mitel VoIP and unified communications applications.

Mitel VoIP products and services are available through Mitel or authorized partners. Mitel can help design and deploy any solution.

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ShoreTel phone system: Product overview Chris Partsenidis, Founder, Firewall.cx

ShoreTel Premises platform is an on-site IP PBX and unified communications (UC) service for the enterprise. The Premises platform has four main components: voice switches, IP phones, ShoreTel Director and ShoreTel Communicator.

The ShoreTel phone system has an interesting approach and, according to ShoreTel, its main advantage is simplified design, administration and high reliability.

The product's core is the ShoreTel Voice switch. These are devices that integrate voice functionality, communication, call routing, IP phone registration and SIP trunking or telecommunications provider termination (ISDN-PRI/PSTN) in one box.

The ShoreTel phone system currently supports 14 different voice switches, each with its own characteristics, providing a wide range of deployment choices and telco connectivity capabilities.

Depending on the amount of IP phones to be supported at a location, customers simply stack multiple voice switches until the desired amount of IP phones has been reached. The highest capacity switch is the ShoreTel Switch 90V, which provides support for up to 90 IP phones, 90 voicemail boxes, up to 12 analog ports and SIP trunking to the Internet service



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provider. Companies that require more connectivity can purchase and stack multiple switches.

Additionally, companies can add redundant voice switches anywhere in their network. These additional switches serve as a backup for the local or remote site. If a primary voice switch fails, the backup switch will automatically take over and users can continue working without disruption. ShoreTel is one of the few VoIP PBX vendors to offer this self-healing feature.

The ShoreTel phone system has an extensive range of IP phones with several features for ease of use. All the phones support Power over Ethernet and have Fast Ethernet (10/100 Mbps) or Gigabit Ethernet ports, depending on the model.

The higher-end models include features such as large color touch displays, conference capabilities, multiple line support, integrated VPN client support and more.

ShoreTel Director facilitates network management ShoreTel Director is a unified management tool that allows for administration, setup, diagnosis and reporting of all ShoreTel equipment. The software can automatically discover and configure any ShoreTel devices and supported services, including voice switches, IP phones, voicemail and call routing.


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ShoreTel Director is installed on either a dedicated server or virtualized platform, and is managed via a Web interface. With ShoreTel's IP PBX solution, a voice switch can be connected at a remote branch and the administrator will immediately see and configure the new device through ShoreTel Director, regardless of physical location.

ShoreTel Director also provides real-time status of the whole ShoreTel phone system. If a problem is detected, a yellow or red light will blink next to the site or device affected. The administrator can then find out more detail about the problem.

If the software's host server crashes, the whole ShoreTel infrastructure will continue to run and function without any interruptions.

ShoreTel Communicator is a user-based application that integrates IP telephony features, messaging, conferencing, video, mobility and other capabilities. The interface allows users to respond quickly to calls and messages, and check online presence of colleagues and much more.

The application supports a variety of users, and depending on users' roles, it allows them to perform a number of different functions.

Pricing for ShoreTel VoIP products depends on a number of factors, including the features chosen. ShoreTel offers a one-year standard warranty that can be extended to multiyear agreements, backed up by its technical assistance center and advanced hardware replacement. More information on support and warranty services can be found on ShoreTel's website.


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Toshiba IPedge VoIP System Product Overview


IPedge is Toshiba's answer for businesses migrating from a legacy PBX (analog/digital) to a pure IP PBX service. IPedge is a software-based PBX that runs under the Linux operating system and is installed on its own dedicated hardware or virtual platform (VMware). The IPedge VoIP PBX series is the next step up from Toshiba's Strata CIX series -- a hybrid PBX that supports both digital telephony and VoIP. The Strata CIX was designed to bridge the gap between legacy and IP-based PBXs.

As with all modern VoIP PBXs, Toshiba IPedge offers rich features designed to make communication within the enterprise as easy as possible. Users can customize various aspects of their IP phone, including address book and ringtones. A full list of functions can be found on Toshiba's website.

Toshiba IPedge server specifications and virtualized offerings Three IPedge editions are currently available, each supporting a different amount of users. The largest model, the IPedge EM Server, can support up to 1,000 users and 440 SIP trunks. However, customers can connect



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multiple IPedge systems to scale beyond this limit. IPedge EM is available as a 1U rack server that packs two Intel Xeon Quad-Core processors clocked at 2 GHz, 12 GB of memory and two 300 GB hard disk drives (HDDs) in a RAID 1 configuration.

IPedge EC Server is a step down from the EM Server and is also offered in a 1U rack server. It supports up to 200 users, up to 96 SIP trunks, and is powered by a single Quad-Core processor running at 2.6 GHz with 4 GB of memory and a 160 GB HDD.

Finally, Toshiba IPedge EP is a compact shelf/table server that can support up to 40 users and 30 SIP trunks. Powered by an Intel Atom Dual-Core 1.8 GHz processor, 4 GB of memory and a 250 GB HDD, it's an ideal compact IP PBX with small space and power requirements.

Toshiba also offers IPedge as a virtual server package, which combines the IPedge PBX with the Toshiba automated call distribution and reporting contact center applications. The all-in-one service consolidates two servers into one, simplifying administration and licensing.

Native SIP trunk support and third-party SIP hardware Toshiba IPedge natively supports SIP trunking, which means, unlike other vendors, no voice gateway is required. SIP trunks from telecommunication providers can terminate directly on the IPedge server. This is a major


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benefit, as no additional hardware is required, therefore, it noticeably reduces costs.

IPedge also offers support for third-party SIP devices, allowing users to select from a range of IP SIP-based products and vendors.

If IPedge requires a PSTN/ISDN connection, a separate voice gateway must be purchased. The voice gateway terminates at the telecommunications provider on one leg and connects to the IPedge PBX on the other, using the SIP protocol.

Customers who already have a Toshiba PBX service based on the Strata CIX PBX system can connect to IPedge in order to expand their coverage and functionality by using Toshiba's IPedge Net product. Customers can also connect analog devices such as old telephones or fax machines using FXS gateways, bridging old technology with the IP PBX.

Toshiba also offers the IP5000 Series phones with Gigabit Ethernet connectivity.

Expanding Toshiba IPedge services and users For companies looking at expanding their IPedge system and services, Toshiba offers a range of applications and services such as IP mobility, a softphone client for Apple iOS or Android, and Call Manager, an application that provides presence status and chat capabilities over the network.




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Toshiba also provides full-fledged automated call distribution for advanced call routing between locations.

IPedge is offered exclusively through Toshiba's dealer network, so potential customers will have to find and contact an authorized Toshiba dealer to obtain customized pricing.

Additional information on Toshiba's IPedge system and unified communications services can be found on Toshiba's website.

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Univerge series NEC VoIP: Product overview


NEC's Univerge SV series IP PBX systems are designed to meet today's business VoIP expectations by supporting commonly found VoIP telephone features, while at the same time providing unified communications (UC) features to enhance productivity.

The NEC Univerge VoIP PBX features various models geared to support different sized businesses. All models support SIP, analog and NEC digital endpoints. SIP trunking is supported across the whole series, while traditional PSTN and ISDN (BRI/PRI) connectivity is also offered, depending on the system's configuration.

The NEC VoIP lineup offers six IP PBX models and various hardware additions to expand connectivity and user support. The wide range of models makes NEC Univerge ideal for both small and medium-sized businesses, as well as large enterprises with thousands of users.

Businesses that invest in NEC's Univerge can expand their system's capacity by purchasing additional chassis and stacking them together. This modular design provides great flexibility and helps keep the costs under control.





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How NEC's Univerge models stack up The Univerge SV8100 is the entry-level NEC VoIP model. It supports stacking to provide a total of 512 phones depending on the phone configuration. A total of 200 SIP trunks are supported, but simultaneous calls are limited based on the available IP resources. The SV8100 is housed in a 19-inch chassis with six expansion slots or, if you're tight on space, a 9.5-inch chassis is available with three expansion slots. Customers can populate the expansion slots with cards that provide additional interfaces to connect SIP trunks, PSTN, BRI, PRI lines, digital NEC phones, analog phones and more.

The Univerge SV8300 is geared toward small and medium businesses supporting a maximum of 1,536 phones, a total of 512 trunk interfaces (PSTN/PRI/BRI) and 512 SIP trunks. These maximum figures are only possible when stacking multiple Univerge SV8300 systems together. The SV8300 is housed inside a 2 rack unit (RU) chassis and offers six expansion slots.

The Univerge SV8500 can support a staggering 4,000+ phones, 1,500 PSTN trunks, 4,000 PRI channels and 4,000 SIP trunks when stacked with multiple identical systems. Customers with increased trunk requirements can also look at the additional trunk chassis available to help expand connectivity.

The Univerge SV9100 is the entry model of the SV9000 series, designed primarily to offer extensive UC features. It, too, supports stacking and


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comes in 9.5-inch and 19-inch form factors, offering support of up to 896 IP phones depending on the phone models and system configuration. The unit supports a maximum of 400 PRI/BRI channels, 400 PSTN trunks and 400 SIP trunks.

The Univerge SV9300 is offered only in a 19-inch form factor with support for stacking and a maximum of 1,536 phones, depending on the phone models and system configuration. The unit supports SIP trunking, PRI/BRI and analog trunk support, with up to 512 trunks of each type available.

The Univerge SV9500 is the largest model currently available and stands out from the rest in many ways. It is the only IP PBX offered in a chassis configuration or a prepackaged rack server (Express 5800 Server) and VMware ESXi virtual appliance. The SV9500 supports up to 4,000 phones per system and a maximum of 1,524 trunks (PRI/BRI/analog/SIP), depending on its configuration and high-density analog and digital interface gateways.

Univerge offers extensive features, management capabilities NEC Univerge IP PBX systems can integrate with Outlook to provide click-to-call functionality and support for mobile devices. NEC VoIP offerings have their own custom softphone application that eliminates the need for a physical phone device and extends the desktop around the world.

The Univerge lineup stays on par with other vendor offerings with the support of voicemail, auto-attendant features, unified messaging, cost




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control tools, advanced operator consoles and many more options for businesses. Licensing is managed using key-activation codes purchased by the business from an NEC partner.

Pricing for the Univerge systems varies and is determined by several factors, including the features chosen. As with most vendors, NEC's Univerge IP PBX can be purchased through official partners, though NEC offers direct sales and support services for large enterprises.

Management of the Univerge series is performed via a Web-based interface, while companies with multiple NEC VoIP systems can use the NEC Enterprise Manager Univerge MA4000 for day-to-day tasks. The MA4000 provides centralized management for all Univerge series products, yet some NEC documents incorrectly note support only for the 8000 series. More information on the product and its capabilities can be found on the Enterprise Manager website.

NEC offers a range of IP, digital, wireless and analog phones designed to cover almost every work environment.

While most of NEC's IP phones have been designed to work seamlessly across all Univerge IP PBX systems, the UT880 and DT800 IP desktop telephones have been specifically designed for the SV9000 series IP PBX, offering increased functionality and capabilities, such as LCD touchscreen (UT880 only), support for NEC UC applications, Bluetooth connectivity options and many more features.

A full list of phones can be viewed on NEC's website.


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