IP Telephony from A-Z - The Complete IP Telephony eBook
Mitel realizes that making the decision to switch to IP telephony and actually deploying the system is no small task.
To help you with pre-decision questions and post-decision planning and implementation, we have created this 77-page IP telephony eBook.
The benefits of IP telephony
- One of the key drivers of converging voice and data networks is cost savings. Money can be saved, with the right IP telephony solution, in almost all areas—from deployment and management time and costs to ongoing toll and lease charges. IP telephony can also help your organization gain a competitive advantage, boost employee productivity, and enhance customer service. However, there are important considerations to analyze when deciding on a solution, including: initial costs such as equipment costs, which include the cost of the infrastructure equipment (voice switches) and handsets (analog or IP telephones or a mix of both); operational startup costs, including the time and resources it takes to plan, install and troubleshoot the solution once it is deployed; and finally, maintenance costs, which includes the cost of labor to maintain the equipment plus whatever costs must be paid to the solution vendor for maintenance and upgrades. This chapter will highlight the benefits of IP telephony and go over the costs in details so that you can make decisions about your deployment.
When you consider what most businesses pay for long-distance, you wouldn’t see a huge need to move to IP telephony, necessarily. Large corporations can be paying pennies per minute for long-distance within the U.S. So while companies beyond North America may realize significant savings on toll charges, these savings are not usually enough to convince a North American company to switch to IP telephony.
Savings for most enterprise networks come from consolidating the voice and data network and using fewer circuits from the public switched telephone network (PSTN). In addition to circuit cost savings, as mentioned earlier, an IP infrastructure requires less time for moves, adds and changes (MACs) and often eliminates the need to hire an outside vendor or service provider to handle them. Moving an IP telephone station temporarily or permanently or adding a new user usually simply entails carrying out a quick and simple GUI-based command. With traditional PBX systems, moving an employee can cost hundreds of dollars in labor. In other words, with IP telephony, each user has their own IP phone profile and the network doesn’t care where anybody is located at any particular time, so MACs are simply a matter of conducting a few commands and can often be easily handled by the user.
With IP telephony, management savings are usually immediate since the information technology team can support the voice network as well as the data network because they’re now one in the same. There is no longer a need to have two teams of technical professionals to handle each entity, which adds up to tremendous savings. Further savings are seen right away when an enterprise needs to make a change, such as relocating an office temporarily in the case of construction. The IT staff simply makes the changes from anywhere on the network (or remotely if need be) and a new temporary office is up and running without outside callers ever being the wiser.
Finally, infrastructure tools like physical ports are no longer needed for IP telephony because physical circuit-switched ports aren’t necessary. There is no finite capacity based on telephony cards in cabinets as IP based systems scale to thousands of users by simply adding additional processing capacity, while sharing a common database. This eliminates forklift upgrades of the past, and preserves investments as a business grows.
All of these cost savings are tremendously appealing characteristics of IP telephony. When you add to them the features that are available for employees, contact centers and receptionists, it quickly becomes obvious that IP telephony is going to continue winning converts.
The added capabilities
Contact centers in many enterprises today are extremely expensive because dedicated buildings are often built to accommodate the many staff members. When a company needs to add additional contact center staffers, traditional PBX-based phone systems must also grow in blocks because ports are bought in groups, rather than scaling seamlessly with each new hire. These factors make contact centers very expensive to maintain and scale. However, with an IP telephony solution, contact centers can grow one phone at a time and contact centers can span several buildings across many states. There is no longer a need for one huge building to house all of the contact center agents. In addition, enterprises are able to leverage expertise across entire organizations, rather than hoping to find a highly skilled team in one location to answer incoming inquiries over multiple communication channels including voice, web-chat and email. With an IP telephony solution, users can sign in from wherever they are (even at home) and is instantly online and available as part of the contact center team.
Another customer service feature available in IP telephony solutions is the hunt group. This feature makes certain that all calls are answered by a live person rather than voice mail, which can be frustrating for callers. With various hunt groups enabled, a call into an organization rings extensions in a specified sequence or rings multiple extensions at once (depending on the company’s preference), ensuring callers reach the person they need without navigating through menus or being forced to wait in a queue.
Remote sites are also easy to bring online. With traditional PBX systems, adding a remote site often requires adding a PBX extender, which can cost almost $1,000 per user for the equipment alone. With IP telephony, again, a user can log in from anywhere and have all the same capabilities as if they were working at headquarters or within the contact center building. With IP telephony, to the outside world, it can seem as though you have contact center locations scattered around the globe to be available 24/7, when really you are simply utilizing IP telephony features such as time-of-day routing and call forwarding to make sure calls are answered quickly by a live human being; these people can be working out of geographically-dispersed branch offices, at remote locations, or even at home. Callers always reach a qualified customer service representative, regardless of what time it is. You are also able to manage peak calling times by having the ability to add other employees, regardless of their location, to the contact center to help meet the overflow demand.
With IP telephony, users can also easily re-route their calls so that they are reached wherever they will be working—they can make these changes themselves, without asking for IT assistance. This “find me” feature also enhances customer service as well as productivity by ensuring a caller reaches the right person, regardless of where he or she might be working. An employee can even program his or her extension to ring based on status—ring through when he or she is in the office, forward to a cell phone when there is no answer, or forward to a colleague when the line is busy.
Employees that choose to bring their own device to work, can use smart phones and tablets as their primary way to communicate. All IP-based solutions enable the same level of functionality with desk phones as with the mobile devices used by employees.
IP telephony savings
- Toll charges – least cost routing avoids toll charge.
- Management costs - System management labor – time and money saved. - Users’ personal profile changes – handled by users, not IT staff. - MACs – quick and easy to handle from anywhere on the network.
- Physical circuit-switched ports no longer required.
- Fewer circuits from the PSTN needed
- Flexibility to enable employees to bring their own devices to work for communications, including smart-phones and tablets.
The customer service advantage
IP telephony offers organizations tremendous customer service value-add. First of all, IP telephony systems provide thorough information right at the time a call comes in by popping data onto an agent’s screen. This information can include the most basic of information, such as caller ID information. By integrating specific business applications with the IP telephony system, more in-depth information can populate the screen, including the caller’s buying patterns, address, current account status, and more. Many IP telephony systems also provide for operators significant background information on the current caller’s experience, such as where the call originated, how many times he or she has been transferred, and whether or not the right person is available to take the call. When the person is again transferred, IP telephony systems eliminate the chance of a caller being asked the same question twice (which is frustrating for callers, and frankly, poor customer service) because the most current information, including notes taken during the present call, populates the next person’s screen.
IP telephony systems also allow organizations to implement skills-based routing, whereby calls are routed via an automatic attendant (attendant prompts the caller to choose from a selection) to the most appropriate agent based on criteria like language, experience, technical expertise, and other details. Advanced features that most service providers charge for are also available “free” with IP telephony, including three-way calling and a built-in conference call bridge. This can further aid in customer service when more resources are required to fulfill a customer request or inquiry, and it also allows conference call access by international parties, a feature most expensive conference call services do not provide.
Finally, IP telephony enables self-service options. For instance, when a caller simply wants to find out information about their own account, interactive voice response (IVR) within IP telephony systems enable callers to securely access that information by providing specific information. This eliminates the need for a contact center agent to take time to answer a call, and it also eliminates the frustration that can occur if a caller is put in queue on hold for the next available agent to find out information that is readily available.
The productivity boosts
IP telephony productivity programs can often transform a company’s desktop application, such as Microsoft Outlook, into a multi-media communications center for integrated messaging, providing such features as directory dialing, contact screen pop, caller ID, call waiting, and calendar integration. Employees have more control over both voice and e-mail messages, in one centralized system, and can forward voice mails to colleagues for improved collaboration and customer issue resolution. IP telephony system reports also keep a history of calls made and received, which is helpful in meeting various compliance regulations. Sophisticated features include on-the-fly document sharing and dial-by-name capabilities. Workers are dialing one another, conferencing, transferring calls between locations, and changing their voice mail preferences all with the click of a mouse. There is no longer a need to call the help desk to make such changes. The bottom line is that employees spend less time navigating complex telephone systems and more time performing critical, revenue-producing tasks.
Soft phones further free people from their desks, delivering telephony capabilities to any PC. With calls directed to a laptop and a headset plugged into the USB port, employees can work from anywhere using their computer and its built-in microphone. Employees who travel a lot appreciate the power and simplicity of a soft phone and customers appreciate not having to dial different numbers to reach someone who is traveling.
The growth factor
IP telephony systems allow for quick and easy scalability to accommodate new locations or growth within existing locations, as well as the ability to add people one at a time as needed, rather than investing in equipment that will handle more than an organization needs at the time. Scalability benefits also work downward: when an organization reduces its staff count, it is simply a matter of removing those users’ profiles from the IP telephony solution. Companies are no longer tied to long leases for equipment that remains underutilized.
Some features available in IP telephony solutions (not comprehensive)
- Business application integration (for instance, tying IP telephony to CRM database)
- Calendar conferencing integration
- Call waiting
- Caller ID
- Click-of-a-mouse simplicity—employees make or transfer calls right on their computer
- Conference call capabilities with onscreen document sharing
- Contact screen pop and comprehensive information about each caller
- Desktop application (i.e., Microsoft Outlook) integration
- Dial-by-name capability
- Features easy to navigate for users
- Four or five-digit dialing to anyone, regardless of location
- Mobility—users can work from anywhere
- Three-way calling
The management ease
The best IP telephony systems have intuitive browser-based management interfaces, allowing companies to manage the entire system—from switches to voice mail, automated attendant, and desktop applications—from anywhere on the network. The best management interfaces make adding a new user a snap and automatically update every switch and directory feature, including the dial-by-name and number attendant and online directory. System updates are also quick and easy, taking an hour or two at the most when vendors release new code.
In addition to managing the system itself, managing users and MACs is simplified tremendously. Employees can make most of the changes to their profiles without bothering the information technology professionals, and for changes that do require further expertise, IP telephony systems make it simple. There is no longer a need to spend time and money on having a service provider come in. These costs alone can save an organization thousands of dollars a month.
In order to have a low TCO (total cost of ownership), deploying a simple-to-use, easy-to-manage solution may be the right choice.
In an independent survey, The Aberdeen Group published a 2012 report analyzing the TCO associated with IP and legacy TDM (Time Division Multiplexing) telephony systems, based on interviews with 485 users of both systems. The report, IP Telephony TCO for SME: Think Beyond Equipment Cost, measured categories for TCO, including capital cost, recurring cost, training cost, and planned versus unplanned downtime.
Aberdeen studied Avaya, Cisco, Microsoft, Mitel, Mitel and legacy TDM systems. Almost 75 percent of those respondents were categorized in the small-to-midsize (SME) sector.
Although Mitel has a slightly higher initial implementation cost than the industry average - $691 compared to $636 - when users were queried further about full accounting of implementation costs, Mitel had the lowest cost.
When adding training, external labor, network upgrades, implementation labor and capital cost together, Mitel’s initial costs averaged $944 per IP extension compared to $1,485 for all others. See figure on the figure below:
Annual recurring costs for maintenance and upgrades have a longer-term impact than initial cost, Aberdeen said. See figure below:
Mitel’s annual recurring cost of $113 was 46 percent lower than the average of $211 of all others, Aberdeen said, examining training and third party management, FTE staffing, software upgrades, software support and hardware maintenance.
Built-in Versus Add-On Features
Aberdeen said Mitel’s lower initial and recurring costs are not based on offering a bare bones feature set. To the contrary, in several important aspects of IP telephony functionality, (softphones, unified messaging,desktop integration and presence), Mitel is more likely to have these features built-in than other systems. See figure below:
Mitel systems’ downtime was among the lowest surveyed, at less than one fourth of the downtime of several comparable systems.
Aberdeen said the study shows the value of looking beyond IP telephony unit cost as the primary comparison metric to include the total cost of initial implementation, recurring cost, bundled feature set and system performance.
Mitel users, Aberdeen said, “are clearly benefitting from a lower total cost of ownership and reduced system complexity.”
Ready to make the switch?
IP telephony is the way of the future, for a number of reasons. First, vendors are no longer investing research and development dollars into legacy TDM equipment. Second, IP telephony has simplified communications for numerous organizations and their positive results are now well-known.
With TDM, there’s no interoperability, transferring between offices is not an option, and employees are often on different voice mail systems so forwarding messages is not possible. With IP telephony, companies instantly improve productivity with robust feature sets such as built-in conference call capabilities, four-digit dialing across locations, contact center capabilities, and integration with desktop applications.
IP telephony has become an umbrella term for all real-time applications over IP, including unified communications, mobility and online collaboration. The prospective buyer should consider all these technologies in an appraisal.
The decision: vendor evaluation and selection
- You’ve made the decision to go with IP telephony after careful consideration, but if convergence is new to you personally and to your organization overall, the decision is likely accompanied by worry and concern about making the right choices. Your choice of technology vendor for this transition, as in any decision, is one of the most important.
Organizations of all sizes are implementing telephony and unified communications (UC) solutions to increase worker productivity, facilitate increased mobility, and reduce the costs of business communications. But selecting the right solution can determine whether companies realize those benefits, or needlessly add cost and complexity.
They want a system that provides a comprehensive, integrated solution that is easy to deploy, easy to use, simple to manage and cost effective to operate. Those seek a system that offers advantages, including stream-lined deployment and management, easier scalability, and a significantly lower total cost of ownership (TCO).
Enterprise decision makers generally have three main areas of expectation that help them choose the right vendor. These are areas you’ll want to consider as you embark upon the vendor evaluation phase.
- Convergence experience, expertise and vision
- Expert, responsive support
- Customer-focused approach to business
Convergence experience, expertise and vision
Look closely at vendors to determine whether or not they are committed to IP-based communications. Have they built their solution as a true IP telephony system, or are they jury-rigging an old PBX-based solution to “look like” an IP telephony solution? Are their solutions built with flexibility, scalability, and longevity in mind? Will you have to completely rip out your old phone system and move to IP telephony in one fell swoop, or has the vendor built its solution with a phased approach in mind for those organizations that need to replace their phone system over time?
Expert, responsive support
When you’re working with a vendor during the early stages of consideration, try reaching their technical support team during off-hours. Do you have easy access to technical support representatives and a full range of maintenance and support services? Have they committed to working closely with you during initial deployment as well as future and ongoing projects? While you’ll almost certainly have quick and easy access to a sales representative and possibly a pre-sales engineer during the evaluation phase, you need to find out how you’ll be treated once you’ve already deployed your system. Is vendor responsiveness just as good for customers as it is for prospects?
Customer-focused approach to business
This area focuses on the vendor’s commitment to your success. Don’t let vendors come into the proposal using a hard sell approach. If they do, they aren’t demonstrating a commitment to your success but rather a commitment to their own success (meeting their quota). For real proof points, ask to see a list of the company’s latest customer installations and ask if you can speak with those customers. If things have gone smoothly, they won’t hesitate to let you talk to a customer in the early phase of their deployment. Don’t settle for just a list of customers that have been using the vendor’s system for years. Call early phase customers and ask them if the vendor is still in close contact with them, calls to proactively find out about the installation, and provides onsite support at a moment’s notice during the deployment.
While many companies vary in why they choose IP telephony, most enterprises have found the most common anticipated benefits as:
- Lowering total operating costs
- Enhancing end-user productivity
- Improving IT organization efficiency
- Reinforcing market differentiation and brand image
Organizations in the past have had few vendors to choose from. According to research and consulting firm COMMfusion, in 2013 there are more than 30 vendors and carriers to meet IP telephony and unified communication needs, whether on-premise or in a host environment. The increase in competition means more innovation and better products from a wider selection of companies. COMMfusion notes that SMB, midmarket, and large enterprises have different needs, and different providers to turn to. In general, the most frequently-evaluated IP telephony system/UC vendors are: Avaya, Cisco, Mitel and Mitel, plus the recent addition of Microsoft to the list. The following section will highlight each of those vendors, but keep in mind that there are at least a dozen more to evaluate, depending on the size and particular needs of your organization.
Avaya offers IP telephony and UC solutions with its IP Office and Avaya Aura solutions. The company offers a range of products, including communication servers, a range of voice and video endpoints, as well as unified communication and collaboration applications, video conferencing infrastructure, and more. According to COMMfusion, Avaya has a broad portfolio of products and product features, although this involves having multiple product lines and clients.
Cisco has used its strength as a recognized network infrastructure equipment leader to expand its IP telephony position. Products include Cisco Unified Communication Manager; Jabber client, as well as communication servers and switches for large and midsized companies, communication endpoints and clients, communication applications, video conferencing and telepresence products and endpoints, and more. COMMfusion notes that Cisco has become a dominant force in the industry for both on-premise and hosted voice and video solutions, even though it’s a networking vendor at heart. Despite recent progress, Cisco’s IP telephony and UC products are often perceived as being overly complex, requiring multiple servers and management tools.
The most recent player in the market is Microsoft, which offers the Lync unified communication solution. While not a traditional IP-PBX, Lync is primarily an instant messaging and presence solution, which also provides call control or voice communication capabilities. According to COMMfusion, there is a fast growing list of Lync deployments, although the number of Lync Voice users is still very limited. The company still must overcome skepticism of its ability to offer enterprise voice, and trend away from Windows.
In addition to the Mitel 5000 Communication Platform for SMBs, Mitel offers midmarket companies the Mitel Communications Director, which is based on a single, cloud-ready software stream that can be distributed or centralized. These products support the Mitel Applications Suite (MAS) and Mitel Unified Communicator Advanced (UCA) UC application. COMMfusion notes that the company offers strong virtualization capabilities, including desktop virtualization, although the company is losing mindshare outside of the SMB market. While Mitel has had a strong focus on virtualization, not all of its channel partners are proficient in implementing and supporting these systems.
Mitel offers end-to-end IP telephony solutions including its Mitel Communicator and its Mitel Sky cloud offering, as well as IP telephones, communication applications, contact center functionality, mobility, and UC applications such as IM, conferencing and application sharing. The company offers a full IP telephony solution and UC suite in an all-in-one appliance architecture. COMMfusion found that Mitel has very high Net Promoter scores and customer satisfaction, no doubt based on the ease of deploying and managing the solutions. Mitel prides itself on its low total cost of ownership and “Brilliant Simplicity.” The company’s challenge going forward is integrating its premises-based and cloud solutions, as well as merging the different channels for these solutions.
Issuing the RFP
If you work with a network integration partner or consultancy, you may want to call on them to help you with the Request for Proposal (RFP). You may also request a sample RFP from any of the vendors you’ll be evaluating, but be careful to go through and make sure the one you use is comprehensive and not skewed toward any one vendor. If you decide to write the RFP yourself, here is an outline on how to go about it.
RFP: from concept to paper
Assemble your RFP team. Be sure and include an IT representative, a budget specialist, and any senior executives in charge of departments that will use the technology extensively (sales, telemarketing, etc.). This team should be briefed on the IP telephony project and should understand what new capabilities such a solution will offer so that they are informed enough to give you an extensive “wish list” for features that will make them more productive.
- Select a project leader. This person should be experienced in networking and IP telephony, if possible, and should be able to answer basic technical questions related to the technology, if not the specific vendor solutions.
- Assess what you need from the IP telephony solution.
- Evaluate the current situation, including costs, etc.
- Identify key goals.
- Review most common product capabilities and decide on the importance of them.
- Determine if there will be training required.
- Estimate the cost of the project.
- Record your requirements, goals, and recommendations in a tentative plan.
- Present your plan to the appropriate organizational leaders (executive management, financial department, etc.). Get their input before writing the proposal.
- Write the proposal. A typical proposal contains:
- A summary of the proposal.
- A statement of what you need – the reason you’re looking for a new solution. Include every capability the RFP team has mentioned—be sure and get input from executives, managers, and staff level employees so that every need is met. Do not overlook the obvious and assume that every vendor provides one specific capability (you know the saying, “do not assume anything”). Conversely, what is missing from the current telephony solution should also be noted.
- A weighted ranking of all of the capabilities and features should be included (see figure 2.1 for a sample weighted ranking worksheet). Be specific in the features/ capabilities list and avoid “buzz” words that each vendor could define differently. If necessary, describe any word that could be misconstrued, such as “availability,” which vendors often define differently. Again, don’t assume. Include every single capability that you need. The list should be exhaustive. In other words, don’t omit “voice mail boxes for every employee” from the list because you assume all vendors provide them.
- A description of how the project will be implemented and evaluated.
- Provide information about your organization and its technology goals.
- Include a project schedule. Indicate when you want the new IP telephony solution in place. Provide details on how you want to implement: in phases, within three months from the date of selection, etc. Be sure to include how you want each phase to be implemented so that you get as much out of your old equipment as possible and extend the life of existing equipment and handsets.
- Provide an approximate budget.
- Conclude the RFP with specific open-ended questions for vendors, such as: – What is your approach to training? Where is training held and how long does it take? Will the price of the solution cover travel time and expenses for your staff to attend if it is offsite? – Is there a guaranteed response time for support calls? How will your system be updated? Is telephone support all that’s covered in maintenance fees or are other things covered? Is there an option for hourly support? How many support staffers are on call 24 hours a day? Does the solution contract come with a support guarantee? – What is your history? How long has your company been in business? How many customers do you have? How many new customers have you signed on in the past year? The past six months? Are there any current merger discussions? – What about customer references—to whom can we speak? Beyond happy customers, ask to speak with the most recent customers. A reputable company should be able to give you references from the most recent three month period. – How are upgrades handled and what are the typical costs involved? Also, ask what the process is for a customer to make suggestions and specifically ask if they can name some features that were a result of suggestions from users. – What kind of “bake-offs” and industry reports mention your company? Ask for references in the form of reputably published reports and articles.
- Submit the proposal to the vendors you’ve selected in your long list. Your integration partner or consultant, if you have one, can help you with this process, or simply e-mail or fax it to your vendor list.
Seeing is believing
The next step, after issuing the RFP, is to closely review the proposals from each vendor. It will be helpful to use a weighted ranking system to score each vendor based on your long list of requirements. First, rank each requirement based on the vendor’s answer to your checklist items. See Figure 2.1 for a sample worksheet.
Next, ask to see a demo and request a sample set-up to test the solution in your office so you can revise the score based on actual experience. Once you have seen a demo or tested the solution, revise your weighted worksheet to reflect your actual experience. See Figure 2.2 for the revised worksheet and score.
Once you’ve collected all of the information and carefully evaluated your chosen vendors, including the four leaders, think carefully about your organization’s priorities in general, and carefully consider the following qualities so you can clearly articulate your requirements in these areas as you approach your final decision.
High system reliability/availability
Do the vendor’s products include redundant components in the case of a failure? Are there ways to reroute calls around a failed switch, for instance? Is there a threshold past which the system’s performance will degrade? Ask for specific examples of each vendor’s system maintaining availability under the harshest circumstances. Ask customer references specifically about how reliable the system is.
Equivalent voice quality to TDM
You don’t want your own customers to call your organization and know right away that you’re using something of lesser quality than a TDM system. Ask the vendor if it’s possible for you to go to a customer site and listen to actual phone calls to evaluate the voice quality yourself. Or ask customer references specifically if anybody knows they are on an IP telephony system or if it is assumed that it is a traditional system. Customers are usually willing to share the downside of the solutions they’ve chosen, as well as the upside.
Make sure that the vendor you choose knows exactly how you will need to scale the system for your specific needs. For instance, if your organization often grows and shrinks during different times of the year or in some other cyclical manner, ask how new users would be added to support your growth needs. Will new hardware need to be added and removed each time you grow and shrink? Or will the system support your needs up to a certain point, regardless of how many times you change size?
Some vendors are known for requiring a full infrastructure overhaul to accommodate the new IP telephony system. Be certain that you can use your existing network equipment with the new solution, and make sure that when you add new gear, you can do so without needing to consider the IP telephony system. IP telephony is only beneficial if it’s truly part of the network and it doesn’t bring you new headaches or worries further down the line.
Full suite of communications features & business-enabling applications
Cost savings, as discussed in chapter one, are not simply a result of toll charge avoidance. Most cost savings come from the additional features that you get with an IP telephony system. Will the system provide value-added services like call history logging, conference call capabilities, document sharing, follow-me features, etc.? Compare the checklist of capabilities of each system. This is not to say you should simply compare how many features, but rather decide on which ones are most important to your organization and come up with the vendors that meet the majority of your requirements. A simple ranking system for each system offering should work well (see Figures 2.1 and 2.2 for an example using a 1-10 rating system).
Ease of implementation/management/maintenance
IP telephony systems should make life easier for the IT team, not more difficult. Because the new system works on the existing network, everything is managed similarly. If management of the IP telephony solution is not straightforward and intuitive, how long will it take your team to ramp up to the point that the system will be supported adequately? It’s imperative that changes be made quickly and easily so that the addition of a new system doesn’t add burden to busy IT personnel. Some of the most important factors of convergence are how it simplifies life and how it saves organizations in terms of management time and money. Does your staff need to train with the vendor every month, and can you afford their time out of the office? How difficult is it to train users on features of the system, and will they be calling for help more often than usual because of the IP telephony implementation? In reality, users should be calling your help desk less frequently with a new IP telephony system. Even employee moves, adds, and changes (MACs) should be simple for either the user or one IT staff member to make within a few minutes. You should also no longer need a service provider to make these alterations for you—this will save you money and time.
Efficient, integrated multi-site networking
You will want to make sure that architecturally, your solution is built around a distributed design. A centralized solution that distributes applications over the network to other sites is inefficient as far as consuming capacity on the WAN. If a vendor is proposing a centralized approach and suggests “simply adding bandwidth” as the way around reliability issues, remember that bandwidth costs are not insignificant and insist on a solution that is designed for optimal bandwidth utilization. Multi-site organizations inherently require a distributed, as opposed to centralized, solution.
Last but not least: ease of use
Another factor you’ll want to consider carefully is ease of use for end users. While you will undoubtedly need to familiarize employees with the system, training should not be cumbersome or lengthy. The IP phones and call control software should be intuitive and easier to use than the analog phones being replaced. Features like on-the-fly conference calling, drag-and-drop call transferring, and the forwarding of voice mail messages via e-mail should be simple for employees, even those who cannot attend training and have to learn the system on their own. You will likely have remote users logging in and using the system, and it will be difficult, if possible at all, to get those people to a training session. In these instances, you’ll appreciate a solution that users can easily navigate so they come up to speed and begin capitalizing on features that enhance your company’s employee productivity and customer service as soon as possible.
The bottom line
The most important things for you to remember during the evaluation process are the main business drivers of convergence. Make sure the vendor you choose is committed to making these perceived benefits a reality for your organization:
- Lowering total operating costs
- Enhancing end-user productivity
- Improving IT organization efficiency
- Reinforcing market differentiation and brand image
The next chapter will explore the IP telephony implementation from beginning to end, starting with research and vendor evaluation and ending with the actual deployment, and will include a helpful timeline for you to use.
Planning the implementation calendar
Roadmap to IP telephony*
|12 Months||Read, learn, and ask (read about the technology, ask experts)|
|10 Months||Head for the Internet (scour vendor websites)|
|9 Months||Call in the vendors|
|8 Months||Demonstration and trial period|
|7 Months||Do an inventory|
|6 Months||Request vendor proposals|
|5 Months||Choose vendor|
|4 Months||Gauge your network’s readiness|
|1 Month||Pilot installation and testing|
|0 Days||Go live|
*This schedule can be accelerated to fit needs. For instance, if your organization decides to move locations and the timing is right to implement IP telephony, this schedule can be altered to fit a three-month schedule.
The road to IP telephony
12 months to deployment: read, learn, and ask
The first step is research. The fact that you’ve reached this chapter in the book indicates you are fairly certain about deploying IP telephony at least sometime in the future, if not the near future. It is best to be making that decision about 12 months before you want to deploy a new phone system, IP telephony or otherwise. At this time, you’ll want to get your hands on as much unbiased research and as many reports from reputable consultancies as possible. Read the research with the goal being to decide if IP telephony is right for you. For now, pass up reports that talk about vendors, and get your hands instead on technology articles, technical papers, industry event presentations given by independent technologists or long-term experts, etc.
The following resources can be helpful in your search for IP telephony information.
- CIO Magazine (www.cio.com)
- Network Computing (www.networkcomputing.com)
- Network World (www.networkworld.com)
- UCStrategies (www.ucstrategies.com)
- ComputerWorld (www.computerworld.com)
Trade magazines and their online counterparts do cover vendors, of course, but you can find unbiased technology primers and overviews. It’s also helpful to read customer case studies about deployments to learn about the experiences of those companies that have deployed IP telephony. Read case studies for technology tips first, vendor specifics second.
After you’ve searched on the Internet and leafed through your stack of technology publications, invest in some time with industry experts and analysts. For lengthy conversations, you may have to invest more than time—research analysts can be hired on a project basis to provide you with valuable information and insights. But be sure to keep a keen ear out for biases because often analysts are paid consultants for specific vendors and because they know the vendor, they’ll tend to reference them more often than others. Keep your questions, at this point, in reference to the technology. Learn all you can from these experts about organizations like your own that have deployed IP telephony, what their specific challenges were, and what the results have been.
10 months to deployment: head for the Internet
After you’ve completed your technology research, visit the web sites of the vendors you’ve heard about. Read about offerings from the industry leaders Avaya, Cisco, Mitel and Mitel. Learn about smaller companies and what the benefits and drawbacks to their systems are. It’s recommended to take and keep good notes so that by the time you’re looking at the eighth vendor and you’ve forgotten which solutions do what, you’ll have detailed notes to refer back to. This is where you want to establish a long list and then whittle it down to a short list.
You’ll read about each solution with your own organization in mind. Jot down questions as you click through vendor web pages. You may get the answer to the question quickly, or it may remain on your list until you eventually meet with the vendor. If your organization has many offices across the United States, for instance, look at solution descriptions with scalability, flexibility, and ease of deployment mentioned early. If your organization rarely changes in size and has a limited number of telephony requirements, look for solutions that offer the basics at a very affordable price point.
Next, create a checklist or table with some common features. For instance, most IP telephony solutions offer standard features like caller ID and three- or four-digit dialing. As you exhaust the common feature list, start adding unique features that matter to your organization. Learn (or try to learn) what differentiates each vendor you’re considering. If you save the differentiation for the vendor presentation, you likely will get a skewed answer to the question, “What makes your solution different and superior?” This checklist is just the beginning and you won’t do anything with it until the RFP phase.
Read articles about each vendor and mark items off your checklist as you determine what each offers. Start with articles that the vendor links to (usually found under headings like “press coverage,” “news coverage,” “case studies,” “success stories,” and “customer solutions” on the website). However, vendors obviously will only highlight their true success stories. Use an Internet search engine to do a little sleuthing your-self—you may find three or four stories about users’ unhappiness with a certain vendor. Dig for the dirt. Use all of this information for your checklist and research notes.
9 months to deployment: call in the vendors
After you’ve looked at your checklist and decided three or four vendors probably offer the best solutions for your organization, invite each of them to come in and give you an overview of their solutions and a demonstration if possible. You will hear a sales pitch, of course, but you may also hear features you hadn’t learned about, or you may hear the names of customer references that have organizational needs like yours. Whenever a sales person drops a customer name, ask for the contact person to speak with after the vendor presentation. If you are told the customer cannot be a reference, (which is understandable—many companies will not speak as a customer reference by policy), ask for a similar customer that you can speak with. If your organization is a bank with 23 branch offices, ask to speak with a similarly sized bank reference. If the vendor is not able to give you even one customer reference right away, take note and be cautious.
8 months to deployment: demonstration and trial period
After you’ve seen each vendor’s presentation (and possibly after you’ve spoken with customer references), inquire about an onsite demonstration and also a trial period. Some vendors, after they’ve shown you how their system works, are willing to deploy a sample set-up so you can test the solution in your office. Some vendors give you just a few days or a week. Often, as the trial period nears the end, you can easily get an extension just by asking. A reputable vendor does not put a deadline on your decision. They want you to be happy with your choice of their solution; an extended trial period is not a huge cost to them.
Crucial tasks -do not skip
- Talk to multiple customer references: insist on recent customers as well as success stories.
- Get each vendor to bring an RFP into your office, in person, to discuss details.
- Talk to colleagues at other organizations that have deployed VoIP (beyond vendor references).
- When you’re close to choosing vendor, obtain equipment for a trial period.
7 months to deployment: do an inventory
Assessing your current network is crucial to a successful IP telephony deployment. There are a number of things to keep in mind and questions you’ll want to answer about the organization’s telephone usage. The following checklist will help ensure you think of everything.
- Determine your business requirements. How will the system be used? How many calls per month (or day) are made out of your office? Are those calls to customers or internal employees? How many offices will you have on a system? Are there remote offices to consider?
- Look at your LAN. What equipment are you using? Do you have an up-to-date network diagram? Is the equipment current or outdated? Are you using Virtual LANs (VLANs) for security or performance issues? VLANs improve voice quality by prioritizing voice traffic.
- Assess your WAN. How much WAN bandwidth do you have between offices? How many home or remote offices do you have and will you need dedicated circuits or will DSL suffice? Consider whether managed IP services are a fit for your organization as an alternative to traditional dedicated circuits.
6 months to deployment: request vendor proposals
If you work with a network integration partner or consultancy, you may want to call on them to help you with the Request for Proposal (RFP). You may also request a sample RFP from any of the vendors you’ll be evaluating, but make sure the one you use is comprehensive and not skewed toward any one vendor. If you decide to write the RFP yourself, chapter 2 of this book includes an outline on how to go about it.
The next step, after issuing the RFP is to closely review the proposals from each vendor. It will be helpful to use a weighted ranking system to score each vendor based on your long list of requirements. Again, see chapter 2 for ideas about creating these checklists and spreadsheets. After you’ve narrowed down the vendors to a short list, ask to see a demo and request a sample set-up to test the solution in your office. Most vendors will give you a free trial period so you can get more comfortable with the system.
Once you’ve collected all of the information and carefully evaluated your short list of vendors, think carefully about your organization’s priorities in general and start talking to customers. Be sure you get customer references that have similar networks and similar business requirements to your own organization. Again, ask to speak with recent customers: It’s easy to give you a list of happy customers. Ask for a list of the most recent customers signed on—within the last three months, for instance—and call them about their experience.
5 months to deployment: choose vendor
After you’ve taken all these steps, created a feature checklist, and determined which vendor best meets your feature/functionality requirements, you should be ready to make the decision. Be sure and ask any remaining questions before you indicate that you are leaning towards that vendor. It is very important to review the vendor’s website, including where they post press releases. If there have been any recent upgrades or new product announcements, ask how customers are responding and call customer references again. This will give you the freshest input, and you’ll be able to make the most educated decision on the right vendor for you.
4 months to deployment: gauge your network’s readiness
By testing your data network’s ability to successfully support IP telephony traffic and discovering potential performance problems before your system is installed, a network assessment helps you plan, design and implement a successful IP telephony solution. The assessment can be administered by the solutions partner or by the vendor you choose, since both have a wealth of experience with IP telephony that they apply to interpreting the test results. Regardless if you use the solutions partner or the chosen vendor, an expert voice readiness assessment is required prior to installing a new IP telephony system across multiple sites.
In order to achieve toll-quality voice, you need to deploy IP telephony over a properly architected network infrastructure - i.e., it has to provide sufficient throughput and meet latency, jitter and packet loss requirements.
Throughput: How much bandwidth you need depends on the how many simultaneous calls your organization has going on, the voice encoding scheme used in the IP handset or soft phone, and the signaling overhead.
Latency and Jitter: Latency is the time it takes for a caller’s voice to be transported (packetized, sent over the network, de-packetized, replayed) to the other individual. Distance and lower-speed circuits can cause delay. Latency that’s too high interrupts the natural conversation flow (you may have spoken with someone using VoIP - you think they have stopped talking but they haven’t-that’s latency). Latency cannot exceed 100 milliseconds one way for toll-quality voice. Acceptable quality voice can go up to 150 milliseconds and participants can still carry on a decent conversation.
Packet Loss: Packet loss results in a metallic sound or conversation dropouts. It’s caused by congestion, distance and poor line quality. Because IP telephony is a real-time audio service using Real Time Protocol (RTP) running over User Datagram Protocol (UDP), there’s no way to recover lost packets. A mere one or two percent packet drop degrades voice quality.
A thorough assessment uses active application traffic across the LAN and WAN in order to reveal what’s going to happen when IP telephony is introduced into the mix. Test agents send a variety of network traffic packets - using different application protocols, packet size, packet spacing and quality of service (QoS) levels. The tests simulate the various types of IP telephony traffic that are likely to occur on a live network. In addition to measuring peer-to-peer traffic, the agents can also generate real-time client transactions against production servers, including communication with IP PBX servers. This comprehensive approach enables the test engineer to pinpoint the source of potential problems and make recommendations for resolution, thus avoiding unwelcome surprises following the implementation.
1 month to deployment: pilot installation and testing
If you have an integration partner or the vendor you have selected works with regional resellers and consultants, call and schedule a time to determine your needs list. If your organization or the vendor does not have an integration partner, get an engineer from the vendor in to help you with this list. With this person (or people), look closely at the current design of your network and make a list of any equipment upgrades or new purchases you’ll need to make in order to optimize the infrastructure for IP telephony.
Update any existing network diagrams you’ll be using. Be sure to label it so you know it is the original (pre-IP telephony). Next, sketch your new network diagram with the gear included. Determine if there is any overlap and if perhaps you don’t need as many switches as you thought. If you’re not working with an integration partner, you may want to invest some money in having a technology expert take a look at your new proposed network diagram. It’s better to make major changes in the planning stage as opposed to after you’ve taken delivery of your IP telephony equipment. An expert can also make sure you maximize your equipment purchase and may make modifications to your diagram that will save you money in the long run.
After you’ve come up with your new network diagram, begin deploying the gear onto a test network. This will not only help ensure the new system works optimally, it will help you get accustomed to the new equipment so other deployments (to other locations, for instance) go smoothly. At the beginning, the test network should not affect anybody’s workday. During the second phase, transition some noncritical employees or departments to the test network. This will help you further test the system in a real-world scenario and also gets users familiar with it.
0 days to deployment: go live
After you have played with the system for a few weeks or months and made appropriate configuration changes to adapt to your entire organization, begin rolling out IP telephony company wide. An installation in phases tends to work best, even if the phases are over one week. The larger your enterprise, the longer it will take and the longer you may need between phases.
After the rollout, it’s imperative that you schedule end user training. You may handle this by department or location, depending on your organization. Vendor representatives are often available to be onsite to provide expertise and demonstrations during end user training sessions. While your choice of solutions will likely be rich in features, these features should also be intuitive to the end user; therefore training should take just two or three hours, as opposed to all day.
Make sure that the team you’ve put together is available for the duration (right through user training), at least on some level. If you’ve chosen a project leader, this is the person who will know all the details, even if he or she is not working daily on all of them. Once you’ve made the switch, so to speak, sit back and start enjoying the benefits of IP telephony.
The bottom line
You want to take your time implement IP telephony. A year may seem like a long time, but the more time you invest up front, the less money you’re likely to waste overall. However, if you do not have a full year, this schedule can absolutely be accelerated—but do not skip steps, just shorten each cycle to fit your needs. The next chapter will go into more detail about reliability and what’s required in order to ensure maximum uptime. Topics to be covered include redundancy, mean time to repair (MTTR), mean time between failures (MTBF), and network and applications reliability.
Ensuring reliability in IP telephony
The most crucial characteristic of a business phone system is reliability. You must pick up the phone to a dial tone, you must be able to successfully place outgoing calls, and calls must effectively reach your organization. This chapter covers varying IP telephony solution architectures, mean time between failure, mean time to repair, network reliability, and application reliability. It is meant to help you dig deeper into the solutions you’ve narrowed down to your short list so that you can choose the one that fits best into your organization and existing infrastructure and provide you with maximum uptime.
How is reliability different from availability?
Usually when reliability is mentioned in terms of a voice system, the reference is generally about hardware. Without hardware reliability, the system cannot be reliable. Reliability is determined by calculating how often the system fails compared to the percentage of time the system is available. In the telephony world, “five-nines” reliability is the acceptable benchmark. This means the system is available at least 99.999 percent of the time.
Availability, on the other hand, is predicted based on the probability of a hardware component failure. It is predicted by taking into account the type and number of hardware components in a system and calculating the mean time between failure (MTBF). So, if an IP switch has a predicted MTBF of approximately 135,600 hours, and each failure requires one (1) hour of mean time to repair (MTTR), we would use this simple computation to estimate the availability:
This demonstrates that this particular unit will achieve “five-nines of availability.” Alternatively, this switch is predicted to be unavailable for one hour every 10 years.
Let’s take a household example. Consider a toaster that works for a year (an average year is 365.2425 days = 8,765.82 hours or 8,766 hours), and then it breaks, so you have to replace it: MTBF = one year. You take it to the store for a replacement the next day: MTTR = 24 (one day).
This indicates two-nines availability. However, if you keep an extra toaster on hand, MTTR could be as little as fifteen minutes (.25 hours). While this increases the cost of equipment, it also increases the availability fairly significantly.
Back to industry terms, there is no ordinary telephone system that can achieve five-nines. Since state-of-the-art MTBF for systems is 100,000 hours and MTTR is 24 hours, you would need to deliver 2,400,000 hours between failures to achieve five-nines.
Even repairing the problem in 4 hours doesn’t make it much easier to accomplish:
You would still need 400,000 hours between failures. These examples are far beyond state-of-the-art. The way to meet these demands is via redundancy. Read on for a section on redundancy and specifically n+1 redundancy.
Distributed vs. Centralized, Chassis vs. Modular
IP telephony systems differ in their architectures: Some are centralized while others are distributed. In a centralized setup, the centralized call control server provides dial tone for all phones, whereas a distributed model is one where end points are handled by multiple call control servers. In this solution, call control is provided by each switch in the system. See figure 4.1.
A classic chassis includes a number of circuit boards, with most of them providing telephony interfaces and one consisting of a specialized computer system, while some modular units contain a single board. The classic chassis can be compared to a string of holiday lights: If one bulb fails, the entire segment fails. The more lights on the string (number of circuit boards in the chassis), the more vulnerable it becomes to failures.
A typical chassis model, because you have to take into consideration the reliability of their components, typically has an MTBF in the 50,000 range, which is four (not five) nines availability. This can be raised to five-nines by adding switches for redundancy (costly but effective). More on this will be discussed in the n+1 redundancy section later.
In contrast, a modular architecture includes small, simple and reliable hardware. This modularity is more reliable and also offers more freedom in the design stages of an IP telephony implementation. Look at both modular and chassis-based systems, but keep in mind your specific reliability needs and remember that modular systems generally make configuration changes simpler and seamless.
The bathtub curve
Electronic product failures historically demonstrate a failure profile known as a “bathtub curve.” See Figure 4.2 for a depiction of the bathtub curve. Because of a number of reasons, including stress, electronics tend to have a short life before they start failing. At the beginning of the lifecycle (the left side of the diagram), manufacturing defects, defective parts, contamination and other factors cause failures, before these settle to a much lower level (the middle of the diagram). The other end (on the right) signifies the end of life or wearing out of the product.
Be sure and ask vendors about their failure rates and how long a product lasts before end of life. If a vendor does not give you a concrete number based on scientific calculations (not marketing hype), ask more questions or talk to someone at the organization who can give you that information.
Mean time to repair (MTTR)
When a product is down, the entire system’s availability percentage is dramatically affected. Consider the following example, where MTTR goes from 1 to 24 hours.
The more complex an IP telephone system, the longer it’s going to take to identify what’s going wrong during a failure. Only when you’ve identified what’s wrong can you get a replacement for it, which can take even more time, and then there is the time it takes to get the system back up and running. Because of this, chassis systems described earlier in this chapter require personnel with significantly more expertise to ensure the system remains functional.
A 4-hour MTTR is industry standard, which creates a problem for IP vendors that want to maintain five-nines of availability with a 4-hour MTTR. Redundant systems are usually added to ensure this availability because a 4-hour MTTR requires a 400,000-hour MTBF to achieve 99.999% availability. (Availability = MTBF/(MTBF+MTTR) = 400,000/(400,000+4) = 99.999%.) Modular, distributed systems tend to make system repair easy, which results in a lower MTTR. These systems only require one power source and two or three cable connections.
Moving parts and complexity
Another thing to keep in mind is the number of moving parts there are in a system. For instance, adding a disc drive (rather than flash memory) with a 500,000-hour MTBF cuts the system’s overall MTBF in half. Moving parts are also likelier to wear out faster than non-moving parts. For instance, the bathtub curve for disc drives is steep and it’s often recommended they be replaced well before end of life to avoid failure. In the case of an IP telephony solution, you’d be replacing a disc drive during the time you have it on your network, since most disc drives last five years. Ask each vendor how many moving parts there are in each system. Again, insist on getting the information from another company source if the sales team does not have this information readily available.
Redundancy also impacts the failure rate, ironically. While vendors often add redundant parts, such as disc drives and power supplies, to their systems, the very fact that the number of parts are being doubled in itself can increase the chance that the system will fail (increase the MTBF). When you are considering an IP PBX system for your organization, be sure to look at how complex each system is. The more complex, the longer it takes to repair because problem diagnosis, part replacement, and system restoration can be difficult. Look for modular systems that are easy to manage and troubleshoot, with specific built-in tools to ensure quick and easy diagnosis and repair.
N+1 redundancy and no single point of failure
Look for a solution with a distributed architecture that allows for the use of n+1 redundancy, which means that extra parts—as opposed to entire units—can be added to provide redundancy. Some vendors have 1:1 redundancy, which means twice the hardware is used to accomplish redundancy. Other systems use n+1 redundancy— which improves reliability since it is not doubling the hardware. For instance, the n+1 redundancy solution may need two extra units (where parts to the IP telephony system are duplicated within the two units), while a 1:1 redundancy solution needs five extra units because each unit is duplicated in its entirety. Essentially, using n+1 redundancy creates a multi-unit system with no single point of failure.
In addition to a distributed architecture that provides n+1 redundancy, look for a solution that interconnects each module using IP rather than cards in box slots. This design uses the Internet as a bus rather than having a proprietary backplane, which allows you to use a wide variety of chips and software and also reduces the costs and increases speed because of the use of IP and Ethernet. This design also allows you to seamlessly scale your system to meet organizational growth demands, just as the Internet allows for growth. Finally, look for a system that provides most of its feature upgrades via software so that there is minimal time between the release and your organization’s use of these features.
The goal of five-nines reliability is impossible for most systems because redundancy requirements can be complex and expensive. Using n+1 redundancy is not only more cost-effective, but it is less complex, which in turn reduces the chance of failure.
The biggest hurdle when implementing an IP telephony solution is ensuring it works properly with the existing underlying infrastructure. LANs and WANs have lower reliability than telecommunications systems and are prone to quality-of-service (QoS) issues that make IP telephony solutions unreliable. LANs have multiple serial components, which negatively affects the reliability (typical LANs achieve three to four nines of availability), but it is possible to achieve five-nines availability on a network by using a redundant aggregation switch with redundant paths. After all, four-nines reliability translates to two hours of downtime per year. Can your organization afford that? Most 24/7 operations cannot. Focus on solutions that allow these redundant paths to an aggregation switch.
WANs cause the biggest headache because WAN links are generally available only 99% to 99.9% of the time, and voice quality availability can be as low as 98%. If your employees depend on superior voice quality for their many conference calls, for example, this is going to be a problem. Some solutions exist that distribute call control to local switches, which means that if a WAN link goes down, a remote switch can handle the calls because call control, business logic and system database information are all available within that switch.
A system with centralized call control relies heavily on its WAN connection because when it goes down, remote sites have no call control, which means calls cannot be made unless a backup system is in place. Look for a distributed solution that provides full and seamless call control functionality even during a WAN failure.
In addition to ensuring your system is reliable in terms of hardware, you must also ensure that IP telephony system applications, including auto-attendants, voice mail, and desktop integration, work all the time for your employees. Look at systems that offer one application server for a full range of applications. You can use more than one server depending on your organization size, but make sure that it is not one feature per server, like some solutions may force you to do. A truly reliable system, in terms of applications, uses a site hierarchy, which means the first application in a user’s hierarchy is used, and each application server has access to the configuration database in a central server. This design is highly reliable because each application server caches the configuration database, making information and applications available even during network downtime. For example, in the case of a network outage, remote users with their own server are unaffected by a failure in another server so that individual sites can serve features like auto-attendant.
The bottom line
There is always the possibility that a system can be completely unreachable because of multiple LAN and WAN problems (remember the saying, “never say never”). Look for solutions that allow you to build into your system a backup plan, such as the ability to implement failover trunks, switch failover, and copper bypass for emergency service. There are lots of vendors out there offering piecemeal solutions that could leave you dealing with increased complexity and decreased reliability. A distributed architecture is a good fit for multi-site organizations, and n+1 redundancy designs will keep your costs— and your chances of failure—way down. The next chapter will go over system handsets, including analog and IP telephones as well as hard and “soft” phones.
Handsets and interfaces
Think about your home telephone, your smartphone, tablet or any one of the multiple electronic devices you use every day. You expect—and appreciate—a well-designed product. You shop for these items with design and functionality in mind. With IP telephony, you can now bring the same high expectations into the office and into your search for handsets. Mediocre office telephones are a thing of the past because IP telephone handsets introduce so many more features and benefits.
Business workers rely on the telephone many hours out of the day, from collaborating with business partners and co-workers to interacting with and helping customers and suppliers. Contact center professionals literally spend the entire day on their telephones. It’s not enough to “make do” with a standard, feature-lacking desktop handset. To make employees more productive—and happier—you need to provide them with the tools they need to do their jobs optimally. You’ll only do this when you present them with a handset that is ergonomically well-designed, has great sound quality, and features a multitude of capabilities at the touch of a button.
Ergonomics is the science of designing products, machines and systems to maximize the safety, comfort and efficiency of the people who use them. Ergonomics takes into account psychology, physical measurement, environment, and more to ensure that products are adapted to suit workers and their specific needs. Keep ergonomics in mind as you look at the handsets and graphical user interfaces (GUIs) of each vendor’s solutions. If your organization is a machine shop, the most important feature for your handsets may be a very loud ringer. If you have a contact center staff, a bevy of features that help shorten the call cycle will be most beneficial. A law firm may require a system that logs incoming and outgoing calls and keeps this information on record for future reference. A recording studio may require ultra-clear sound quality to ensure recorded voices are pitch-perfect. Look at your organizational needs in terms of what you need a handset and GUI to do for your employees.
IP telephony, with its packet-based design, is able to deliver better than toll-quality sound with hi-fidelity audio and innovative design. Better sound translates into productivity gains – shorter calls with fewer errors, increased sales because of the clarity of conversation between a sales person and customer, and increased caller satisfaction. Wideband audio is preferable over narrowband, because it has an increased range on the low end (50-300 Hz) and makes conversations sound less tinny and reduces error in translation. Look for a solution that supports both wideband and narrowband.
Speakerphone microphones are also an important part of sound quality consideration. Look for a solution that supports hi-fidelity sound and has a full-duplex operation speakerphone so audio flows freely on both ends (no delay if one speaker talks over another). Not all IP speakerphones are able to do this. In addition, ensure you choose a handset that meets the Americans with Disabilities Act (ADA) regulations for the hearing impaired, regardless of whether you have an immediate need or not. (More on ADA compliance will be covered later in this chapter.)
IP telephones act more like computers than telephones—they have a bigger screen and more functionality attached to the screen. This screen also delivers more information about each call and prompts the user through the call with various options appearing on the screen. The user simply presses a corresponding key below the screen to accomplish any task while on the call (call forward, conference, etc.).
Make sure to consider carefully the size of the screen, with your users in mind. Is it big enough that after a long day of work, it’s still pleasing to the eye? Is the display big and bright enough to see clearly after four hours on the phone? In addition, work with the phone and test what features are available and how easy those features are to access for a contact center worker taking up to 50 calls an hour. Is there a message waiting light to ensure no message is missed?
Another characteristic to consider is the feel of the phone, since that is another source of fatigue for users. The phone should minimize shoulder and neck pain and fatigue, and it should essentially fit most users comfortably. The handset should not be too light or too heavy—try and get a phone with a balanced weight of about 170-190 grams. Also, consider a handset with a grip that is covered with a smooth rubber material, as opposed to the slippery plastic kind that can become uncomfortable during long telephone calls.
Many systems will come with fixed-feature keys. Make sure those features are the one most pertinent to the needs of your employees. Fixed features usually include transfer, conference, intercom, voice mail dialup, directory, and redial. If a system relies mostly on soft keys, consider how difficult it may be for all your users to get those soft keys set up and working. Will you end up having to go around to every employee’s phone to program two or three soft key functions? Soft keys are beneficial to have, but they should not be used for standard functions—these should be on hard keys. Some functions that you should look for on fixed-feature keys include:
- Directory: This key should be linked to a quick-dial program that allows a caller to dial by name using the telephone keypad (7 for S, 2 for A, 6 for M, which would bring up names that match beginning letters “SAM”).
- Redial: This function key should do more than simply dial the last number dialed— it should allow you to press it and see an historical list of outbound, inbound and missed calls.
- Personal options: This feature key should allow for easy management of personal options, such as ring tone and call handling preferences.
- Voice mail: This key should provide quick and easy access to voice mail messages.
Soft keys are multi-function keys that use part of the telephone display to identify their function at any moment. They are usually located directly underneath the display and their use changes depending on where the user is in the call process. You can set some soft keys for use by all of your employees, and you can choose to leave some to the discretion of each user. Make sure the setup of soft keys is straightforward before allowing users to set up their own. If the IP telephony system you’ve chosen does not offer handset soft keys that are easy to set up or change, make sure the solution allows you to either set the soft keys for each user (or block users from trying to set up their own) or the ability to choose not to use the soft keys at all. This will minimize user confusion and frustration if the solution is difficult to edit.
Business vs. basic phones
Basic phones differ greatly from business phones in that they offer few or no additional functions beyond answering and hanging up. Business phones streamline tasks and offer users productivity enhancing features. You’ll find that some vendors offer most functionality via soft keys, while others rely on numerous hard keys—one function per key. Your employees may fare best with fixed function keys, or classic business phones, which generally have a button per task. Some vendors do not offer this, however, relying mostly on soft keys. Still other phones offer a fixed number of hard keys and some extra hard keys that you can program to fit your organizational needs. These are optimal for organizations with workgroups that need specific functions to be programmed into keys.
Easy to manage
You want to make sure the phones you are getting with the IP telephony system you choose are plug-and-play, particularly if you have a large organization with many locations, some of which have no technical staff on-hand for installation support. Nontechnical employees should be able to plug in their phone and start working. When it’s plugged in, the phone should automatically get its IP address, subnet mask, and gateway, as well as the accurate time from a time server. Handset updates should be equally as hands-off for employees—updates should be automatic as they are released by the vendor.
While most businesses do not place emphasis on how a phone looks over the functionality, it is still an important consideration. A phone that is pleasing to the eye is as impressive as a beautiful desk or sleek-looking computer. Consider your options with your chosen vendor and ask about variety. What colors do their phones come in? Are there smaller versions for users who need minimal functionality? Are there ruggedized versions of the IP phones for public area usage? Look for a solution that will fit all your needs, with phones that are consistent in appearance and look classy throughout your organization.
In an IP telephony solution, the IP-PBX manages telephones throughout the enterprise and acts as a gateway to both voice and data networks. Any kind of telephone, whether it be analog, IP or a soft phone, can connect to the IP-PBX via the network and calls are routed via the network instead of the public switched telephone network.
A regular analog telephone, the same ones you’ve been using throughout your organization until now, can be used in an IP telephony solution to input the caller’s voice into the system. Once in the system, a series of analog-to-digital conversions and other processes change the voice signals into data, which is then transmitted over the LAN, WAN, or Internet. The voice data is then converted back into sound by the recipient’s phone. Most IP telephony systems will allow you to use your existing analog telephones with the solution—forever or until you are able to afford and/or replace them with IP telephones. Be sure that your vendor will allow you to phase out older analog phones with their IP phones over time so you can maximize your existing equipment.
IP telephones (or IP endpoints) actually perform the analog-to-digital and/or digital-to-analog conversions and can plug directly into the LAN or WAN. VoIP system vendors usually offer a variety of IP telephones so that you can choose different models based on various segments in your user population. Your legal department may need multi-line handsets with easy conference call capabilities. A manufacturing floor needs a phone with fewer bells and whistles but good, loud sound and a rugged exterior. Receptionists need handsets with many more fixed feature buttons so that they can handle calls quickly and accurately.
A soft phone is essentially software that is used to make calls over an IP telephony system using a personal desktop computer and either a headset connected to the computer’s sound card, or a telephone connected to the computer using an adapter. It behaves like a traditional phone but usually offers much more information to the user, depending on the vendor’s GUI. When a call comes into a station with a softphone, an icon appears on the computer screen, which allows the user to either answer it by clicking on an icon, or ignore the call by clicking on another icon, which in turn sends the caller to either voice mail or another employee.
Often, vendors offer an application that allows traveling employees to gain access to the robust feature set of their desktop computer from wherever they are working—at home or on the road. A user simply logs into the system from the local phone and has access to all of the same functions he or she would enjoy while in the office.
WiFi phones use signals much like those used by cordless telephones. The WiFi phone receives signals which allow you to wirelessly connect to the network via wireless access points (APs). Unlike traditional cell phones, the technology of WiFi phones allows them to transmit data at really high speed, but areas of coverage are limited by the reach of the AP being utilized. There are also hot spots available in various locations (restaurants, Starbucks, libraries, etc.) that allow you to access the Internet using your own WiFi service (or a service utilized by your organization).
This accessibility contributes to the BYOD (bring you own device) trend as workers more easily access their corporate networks with their phones or tablets via WiFi.
In fact, mobile devices from Apple and other companies are fast replacing PCs in the enterprise, as workers fuel the BYOD phenomenon. Workers want to use their favorite mobile devices in the workplace and they are asking IT managers to help them integrate them.
One drawback to WiFi phones is the fact that some things can impede on the quality of the calls, such as how many people are using the same hot spot, how close the WiFi phone user is to the access points, WiFi card capabilities, and possible obstructions to the AP (such as a wall). Another drawback is that WiFi technology does not offer the level of security offered with standard Internet access. More on security will be covered in the following chapter.
Want a SIP?
Session Initiation Protocol (SIP), a signaling protocol, is used for establishing a session in an IP net-work—from a simple two-way telephone call to a multi-media conference call session with many participants. The IP telephony industry has adopted SIP, an RFC standard (RFC 3261) from the Internet Engineering Task Force (IETF), as the protocol of choice for signaling because of its ability to facilitate Internet applications by working with other protocols. It is not the be-all and end-all of protocols—it was designed to be a facilitation mechanism, not an all-inclusive solution. Its flexibility is what makes it so powerful, and an all-inclusive approach does not offer this level of flexibility.
Essentially, SIP establishes, manipulates and tears down sessions, and its main purpose is to help session originators deliver invitations to potential session participants wherever they may be. It uses URLs to address participants and SDP to convey session information and it’s easy to combine SIP with other applications, like Web browsers and messaging. The bottom line is that it’s a modular approach to maximizing IP telephony protocols. SIP can find and invite call invitees wherever they are. It facilitates multi-media calls with many participants who may join and leave at will.
American Disabilities Act (ADA) compliance
Your IP telephony system must comply with the American Disabilities Act (ADA) of 1990 and associated regulations issued by Federal agencies that define guidelines for accessibility by individuals with disabilities. These guidelines include requirements for telephones and telephone systems, and they include the “ADA Standard for Accessible Design” (Pt. 36, Appendix A, Section 4.31, Telephones) and the 508 provision for TDD/ TTYs. A few of these requirements include:
- Volume Control: Telephones should have volume controls that provide a gain adjustable up to a minimum of 20 dB. The telephones should provide at least one intermediate step of 12 dB for incremental volume control.
- Automatic Volume Reset: The telephone should automatically reset the volume to the default level after every use.
- Hearing Aid Compatibility: The telephone must have a means for effective magnetic wireless coupling to hearing technologies.
- Minimized Interference: Interference to hearing technologies, including hearing aids, cochlear implants, and assistive listening devices, shall be reduced to the lowest possible level that allows a user of hearing technologies to use the telephone.
- Support for TDD/TTYs: Products that transmit or conduct information or communication shall pass through cross-manufacturer, non-proprietary, industry standard codes, translation protocols, formats or other information necessary to provide the information or communication in a usable format. Technologies which use encoding, signal compression, format transformation, or similar techniques shall not remove information needed for access or shall restore it upon delivery.
- Controls and Keys: Controls and keys shall be tactilely discernible without activating the controls or keys. These controls and keys shall be operable with one hand and shall not require tight grasping, pinching or twisting of the wrist. The force required to activate controls and keys shall be 5 lbs. maximum. If key repeat is supported, the delay before repeat shall be adjustable to at least 2 seconds. The status of all controls or keys should be visually discernible, and discernible either through touch or sound.
- The cord from the telephone to the handset shall be at least 29 inches (735 mm) long.
- A wall-mounted object should not protrude into the walkway more than four inches to ensure visually impaired individuals do not run into them.
The bottom line
By now, you have either chosen your IP telephony vendor or at least narrowed it down to a short list. Take the telephone characteristics into account to help you finalize the decision. If you have already made your choice, look carefully at all of the models your vendor offers and choose the right phone for each user in your organization: Multifunction telephones for receptionists, soft phone licenses or WiFi phones for travelers, basic but ruggedized phones for warehouses and manufacturing floors. At this stage in the IP telephony game, you have more options than ever and you don’t need to make one model work for everyone. What you do need to do is make sure your users are more productive because of the phones, and that your choice complies with the ADA. The next chapter will cover how you can secure your IP telephony communications.
Anybody who’s connected to the Internet or who owns a tablet/multi-function smartphone knows that they’re at risk of getting viruses, worms, spam and other malicious threats. In addition to the potential damage these threats introduce in terms of lost data or corrupted files, there are now regulatory issues associated with ensuring protection. Healthcare has its own privacy regulations in the form of HIPAA (Health Insurance Portability and Accountability Act of 1996), and infringements can result in significant punishments and fines. The bottom line is that you have to protect your organization’s devices and network. IP telephony is no different – the only difference is the form of the traffic: voice versus data. All traffic crossing a network can be stolen, manipulated or blocked if proper network security precautions are not put into place. This chapter will highlight the steps you should take to ensure your IP telephony traffic is secure against outsiders and unauthorized individuals.
Evaluate your risks
The first step to determining the right network security strategy (VoIP or otherwise) is to determine the risks your particular organization faces. (Because of the increasingly complex network security threats and solutions out there, you may want to get a network security expert on board to help with the assessment.) For instance, a healthcare organization faces different regulatory requirements than a legal or accounting firm. An e-commerce organization has altogether different privacy and security requirements. Once you determine what your risks are, you’ll be better able to determine the best multi-layer defense against attacks, eavesdropping, service theft and other evolving threats for the entire network, including the IP telephony system being utilized.
IP Telephony – specific considerations
Network-based attacks. IP telephony is susceptible to Denial of Service (DoS) attacks because these can cripple the network to the point that nothing, including voice calls, can get through. (It is generally recommended that every organization using IP telephony have backup telephone lines in the case of an out of control DoS attack or regional power failure.) Spam, spyware and phishing are other network attacks that are commonly used to commit identity theft and other fraud. Finally, viruses and bots can destroy data or devices or even hijack phones into a toll fraud scheme.
Phone service theft. A hacker could enter into an unprotected network and access the PBX to make endless international calls. There have been major cases cited in the news where toll fraud has cost companies millions of dollars. In many instances, the criminals have been caught and prosecuted, but not without major costs to the companies defrauded; and keep in mind, there are always those crimes that go undetected.
Eavesdropping. Without the proper security in place, a hacker could eavesdrop and possibly expose confidential information. A private conversation about financials could be recorded and played for anybody, which could lead to internal and external problems, including punishment from numerous regulatory agencies. Or a personal call from an employee to a florist with a credit card number could lead to credit card and even identity theft.
Power failures. While outages affect data traffic, of course, there’s a difference when it comes to telephony. People expect telephones to work even during an outage because homes often have a nonelectronic phone that simply plugs into the telephone outlet. This expectation is generally brought into the workplace.
SPIT. Spam over Internet telephony is an alternative to telemarketing where one message can easily be sent to thousands of recipients with the click of a mouse. In other words, your employees’ voice mail boxes can become as overloaded with spam as their e-mail would be without appropriate spam filters.
Other threats. There are new threats created and discovered daily. One such attack is the spoofing of a phone number, which essentially allows a hacker to look like he or she is someone else, which is one of the easiest ways for this person to steal an unsuspecting person’s identity. While individuals have learned not to trust e-mail, it is still generally believed that telephone communications can be trusted.
Network security basics
Network security will lead to a secure IP telephony system. Your organization has likely taken steps such as initiating the use of virtual private networks (VPNs) and installing firewall equipment, which protects the organization against intruders and threats mentioned earlier. Since voice is just another application on the network, the same precautions should be taken to secure the IP telephony equipment. Every form of security should be applied, including physical, human, network, and system security.
- Physical security: Buildings, equipment rooms, data servers, and wiring closets should be off-limits to anybody who is not authorized.
- Human security via security policies: Make sure your organization’s informational assets are protected against inappropriate or unauthorized use by a renegade employee. Ensure hiring and system usage policies are in place to govern appropriate use. Establish and strictly enforce policies having to do with passwords and system usage.
- Network security: Again, create a multi-layered defense using firewalls, VPNs, and intrusion detection or prevention (IDS/IPS). Make sure wireless access points use the highest level of access control and encryption to prevent intruders from gaining access to your network and its resources.
- System security: Arm every desktop with anti-virus software to fight against spyware and other malware. Utilize host intrusion prevention systems to protect servers against attacks.
Another force to consider is segregating traffic via virtual LANs (VLANs). It is a method of logically grouping devices or departments onto their own LANs. Isolating LANs from one another provides an additional layer of security. It also reduces the impact of multicast or broadcast traffic since there are separate broadcast domains.
Finally, bandwidth management can be utilized to further guarantee bandwidth for business-critical, latency-sensitive traffic like VoIP traffic. Bandwidth management methods include assigning a certain priority to each type of traffic. VoIP packets should be assigned the highest priority to ensure voice traffic gets through.
IP telephony security basics
When your network is secured, take it a step further and utilize best practices for deploying secure IP telephony.
- Firewalls: Make sure the firewalls you’re using can handle the latency sensitive needs of IP telephony traffic.
- Switched environment: Use Ethernet switches (not hubs) to connect all your voice devices not only for better performance but also to limit the possibility of a hacker getting onto a call because in a switched environment, the flow of traffic is between devices and nobody can tap in.
- VLAN assignment: Assign voice to a separate VLAN (or separate VLANs). This segregates traffic for improved performance and security.
- Priority: Prioritize voice traffic over data on these VLANs so that delay sensitive traffic gets through even during a network attack. Ensure your network switches can prioritize based on VLAN tags and support multiple queues.
- VPN: Use a VPN between sites, buildings, or departments to encrypt traffic. This is especially important when it comes to protecting confidential employee information, such as social security numbers. In addition, use software VPNs or VPN appliances for remote users to protect conversations from being tapped. Your system should also offer you the option of completely disallowing remote access for an even tighter security option.
- Port lockdown: Lock down IP telephony traffic on the physical switch ports so that only authorized MAC addresses can transmit over the port.
- Media encryption: Look for a solution that prevents eavesdropping by encrypting voice traffic. This way, even if someone taps a voice stream, they are unable to decode or understand the conversation. Not all IP telephony system vendors offer this but it is a necessity for IP telephony security.
- Voice mail storage: Make sure that your voice mail storage is itself secure to prevent unauthorized access of voice mail files.
IP telephony system security
Let’s look now at the IP telephony system itself. While you can secure your network in all the right ways, you also need to choose a phone system that is secure itself. Consider moving away from a system that uses Microsoft Windows for call control because of the security considerations. With a constant stream of Windows security updates and patches, you’re risking downtime and security breaches.
Another architectural consideration to keep in mind is ensuring your system is distributed, which will mean it has no single point of failure. A distributed system allows continued operation in the case of worms, viruses, or DoS attacks. An attack will not disable the entire system if intelligence is distributed amongst multiple devices.
Your chosen system should offer multiple levels for administrator permissions to limit control and ensure unauthorized individuals do not gain access. Once you’ve deployed, reserve full access for just a few key information technology employees. Ensure that a web-based management solution supports secure management using Secure Sockets Layer (SSL), which secures communications from the interface to the server.
Best practices for securing IP telephony
Iron-clad IP telephony security is built on top of strong network security. Here are best practices for securing IP telephony in the WAN, the campus and local networks, and for remote users working from home or the road.
Best practices for deploying secure IP telephony over the WAN include:
- Use a VPN Between Sites. When interconnecting multiple locations, organizations may use managed networks, point-to-point communications or an IP service provider. Whatever WAN connection you choose, use VPN tunnels between locations to encrypt communications.
- Use Firewalls. Use a firewall to protect your internal network from the threats coming in from the WAN and public Internet. Make sure the firewall has the performance to handle the real-time needs of VoIP traffic. Specifically, the firewall must be able to handle a large number of small packets without introducing a lot of latency. Mitel has done interoperability testing and has certified Juniper/NetScreen and SonicWall firewalls.
Best practices for secure IP telephony in the local network include:
- Use Ethernet Switches. Use Ethernet switches for your all voice devices, including IP phones, SoftPhones, ShoreGear voice switches and ShoreWare servers to reduce the possibility of snooping into the voice traffic. In a switched environment, traffic flows between the two devices and cannot be observed by non-malicious users. Do not use Ethernet hubs, as it is easy to observe traffic on this shared resource.
- Put Voice in Separate VLANs. Organizations can set up separate VLANs for voice traffic, which eliminates broadcast domains and segregates traffic for improved performance and security. Using VLANs can limit the number of ports for which voice traffic is destined, adding to security. With Mitel, VLANs IDs can be set automatically using DHCP, which saves time. Mitel phones also support Link Layer Discovery Protocol (LLDP) which is an open standard method to assign VLAN tags at Layer 2.
- Prioritize Voice Over Data (LAN). The VLAN can be used to prioritize voice over data on the local area network, which can allow the voice traffic to get through even when data traffic is intense — including some network attacks. Check your network switches to ensure they can prioritize based on VLAN (or DiffServ) tags and that they support multiple queues.
- Prioritize Voice Over Data (LAN/WAN). DiffServ should be used to prioritize voice over data on the LAN and the WAN to ensure the voice traffic gets through even when data traffic is intense—including some network attacks. Check your WAN access devices to ensure they can prioritize based on DiffServ and that they support multiple queues.
- Rate Limiting. Critical network elements like routers and switches should use rate limiting to make sure a single traffic flow cannot consume the entire resource, such as CPU, memory, or bandwidth, in the face of a DoS attack.
- Port Lockdown. For stronger security, companies can lock down VoIP traffic on physical switch ports so that only devices with specified MAC addresses may transmit over the specified port. This process is labor-intensive but it can mitigate local threats. Some Ethernet switch vendors have software to automate this process.
- Prevent Eavesdropping. A malicious employee or intruder who has penetrated your network can use snooping tools to capture a session before the call is initiated and play back the communications later. Attackers can fake the MAC address of a client, pretending to be a legitimate device, and gain access to the network. With port mirroring, all the traffic on one switch port is simultaneously sent to a network analyzer connected to another port. An intruder can use then snoop the network traffic.
Mitel system supports 128-bit AES media encryption which is the ultimate protection against electronic eavesdropping and replay attacks. Even if someone successfully taps the media stream, they cannot decode and understand the conversation.
Best practices for deploying secure IP telephony on campus networks, on building floors and in workgroups include:
- Use VPNs Between Buildings. Use a VPN between buildings on a campus or floors in a building. Because the traffic is encrypted, the information inside the VPN tunnel is protected from eavesdropping.
- Use VPNs for Departments. Deploy VPNs between key departments, such as human resources, finance, executives or legal, whose conversations are often company confidential.
- Use Encryption for Important Individuals. Use the media stream encryption feature to protect communications for extremely important users like generals or CEOs. Media stream encryption is more cost effective and simpler to deploy than VPNs.
Best practices for deploying secure IP telephony to remote workers include:
- Use Software VPNs for Soft Phones. Employees working from home or on the road connect to the corporate network over untrusted connections, be it cable or DSL from home or from a Wi-Fi hotspot at the hotel or coffee shop. Remote workers using the Mitel IP SoftPhone should use a software-based VPN.
- Use Built-in SSL VPN for IP phones. Remote workers using IP phones should use the built-in SSL VPN capability available on the 230g, 560g and 565g telephones. These phones feature the industry’s first built-in SSL VPN client (not IPSEC) that terminates on the ShoreGear VPN Concentrator. Unlike IPSEC tunnels that are typically blocked by firewalls, SLL tunnels look like web traffic and pass through corporate networks. This flexibility allows the phones to be deployed even at shared or temporary office locations.
The bottom line
IP telephony requires the same level of security as your data network requires. You need to ensure you’re receiving calls from trusted sources, you’re protecting your infrastructure from toll fraud, and you need to make sure your voice calls get through, even when parts of the network might be bogged down by DoS attacks, viruses, or worms. There are vendors that offer IP telephony solutions with additional layers of security. You don’t have to rely solely on network security devices in place. You can take it a step further and protect your IP telephony equipment so that voice communications and resources are as safe as possible from hackers and other criminals. The next chapter will discuss wireless IP telephony, including more on security, as well as QoS, reliability, and coverage areas.
Mobility and wireless
In addition to cost savings, productivity improvements, and customer service enhancements, another driving force behind IP telephony is mobility. Workers are increasingly mobile—from traveling sales people to contact center staffers who work from remote sites and even home offices around the globe to serve customers 24/7.
Mobility is an absolute necessity, as is the requirement for customers to reach anyone at anytime, anywhere. IP telephony is the ideal way to meet this need. With it, organizations can use distributed hunt groups to ring employees around the globe with the right skill set to ensure a question is answered or an issue is solved immediately. As long as an agent with the skill set is logged in, even if on another continent, the issue will be resolved just as if he or she were at their own desk.
With IP telephony, calls are intelligently routed based on calendars, so agents logged as out of the office are reached via cell phone, etc. At the same time, the agent’s smartphone acts as an extension of their desk phone with all of the integrated features, such as dial-by-name, transfer, conference call capabilities, etc. This wireless integration is crucial, especially since you don’t necessarily need to purchase specific wireless handsets or specialized handsets for traveling employees.
This mobility is not even noticeable to the customer base. There are IP telephony vendors that allow employees to choose their device—for instance, a smartphone, home phone or tablet —and that device assumes the identity and capabilities of his or her regular office extension. For example, the caller-ID information provided when the employee makes a call can reflect their office number instead of the mobile or home-office phone actually being used. In other words, caller-ID will indicate that the call is coming from headquarters of their company. This is important to protect the employee’s privacy and strengthen the corporate brand.
The continued growth of the BYOD (bring your own device) trend is driving a surge in the adoption of smartphones and tablets in the enterprise. Employees seeking to use the devices they know and love in the workplace are pushing their IT managers to integrate the devices with their corporate systems.
Recent data from global market intelligence provider IDC highlighted the tablet’s rise in popularity, noting that sales in 2012 were up 78.4 percent over 2011, with a total shipment volume of 128 million units. Furthermore, IDC analysts predict tablet sales will surpass sales of desktop PCs and portable PCs in 2013, reaching 190 million units.
While tablets are valuable for a wide range of work purposes, one of the main uses is for communications, whether messaging, phone, document sharing, video conferencing or other activities. The leading tablet, the Apple iPad, can be used for external communications to provide exceptional customer service as well as for collaboration solutions to boost internal communications. In fact, with Mitel’s unified communications services, including Mitel Mobility, the iPad can make use of VoIP (voice over internet protocol) technology to take advantage of affordable calling.
Mitel provides three offerings that fall in line with enterprise iPad adoption: Mitel Mobility, Mitel Conferencing and Mitel Dock. These releases were designed to help optimize business use of the Apple iPad, turning it into a communication tool by enabling integrated collaboration capabilities and increased accessibility no matter where the user is located.
Mitel Mobility provides a native iPad user interface that enables multi-modal communications. With a swipe of the finger, users can place and receive calls with a business persona, send and receive instant messages, listen to voicemail and create multi-party calls by simply dragging names from enterprise directories together.
Mitel’s Conferencing for iOS builds on the communication experiences of Mitel Mobility, adding in application collaboration capabilities. With conferencing solutions, users can use their iPad or iPhone to control or participate in presentations and easily share them with remote conference participants. Mitel’s solution additionally includes desktop sharing functionality, which facilitates collaboration.
Mitel Dock is an enterprise-grade docking station that can be used to transform a user’s iPad and iPhone into a desk phone. An employee’s iPhone or iPad can be placed into the Mitel Dock to provide the user with instant access to the advantages of a business desk phone such as comfortable use, renewed battery life and high-quality calls.
With IP telephony, users are highly mobile, logging in from anywhere and gaining access to all the same capabilities as if they were working at headquarters, at their desks, or within a contact center building. With IP telephony, to the outside world, it can seem as though your organization has contact center locations scattered around the globe, making help available 24/7. In reality, you are simply utilizing IP telephony features such as time-of-day routing and call forwarding to make sure calls are answered quickly by a live human being. Your employees can be working out of branch offices, at remote locations, or even at home. Your workers are mobile and happy; your customers are being catered to and satisfied quickly. You are also able to manage peak calling times by having the ability to add other employees, regardless of their location, to the contact center to help meet the overflow demand.
With IP telephony, users can also easily re-route their calls so that they are reached wherever they will be working—they can make these changes themselves, without asking for IT assistance. This “find me” feature also enhances customer service, as well as productivity, by ensuring every call reaches the right person, regardless of where he or she might be working. An employee can even program his or her extension to ring based on status—ring through when he or she is in the office, forward to a cell phone when there is no answer, or forward to a colleague when the line is busy.
Once you’ve deployed IP telephony on your network, you’ll almost certainly begin to consider how you might add wireless to the mix. With the broad adoption of Wi-Fi networks based on IEEE 802.11, your employees will also inevitably ask you when you’ll be offering them mobile IP telephony since they’ll quickly grow accustomed to the productivity-boosting and time-saving benefits. With wireless, employees can take these benefits beyond the wired network.
With wireless IP telephony, employees are not tied to their desks and delays are further reduced. Consider, for instance, the case of the sales representative meeting with the CEO. While in a meeting, urgent calls can follow him or her to a wireless handset. Take this example into a hospital, and it can mean the matter of life and death if a nurse is visiting a patient whose health suddenly degrades. The nurse need not waste time running to the nursing station to call the doctor or paging for help but rather, he or she can call the doctor directly from within the patient’s room from a wireless IP handset, provide information and take steps the doctor is advising all in real time as a result of the phone consult.
On top of the savings offered by IP telephony, going wireless can also save your organization additional money. For example, when an employee is working in another location other than his or her office, calls can still find that person if they are free to talk, thereby eliminating any toll charges that would have been associated with returning a missed call, had a caller gone to voice mail.
Prepping for wireless
Until now, many companies have used proprietary wireless voice systems for their warehouses and distribution facilities, for instance, but today, there are standards in place—namely, Session Initiation Protocol (SIP)—for call control over wireless LANs (WLANs). There are other requirements your network must meet, such as sufficient wireless coverage, network scalability, Quality of Service (QoS), and seamless roaming, and of course, security.
You don’t want your users hitting dead zones while they’re in the middle of a conversation. It’s poor customer service and costly to your business. Assess how many users you have in each location of your organization, and consider the bandwidth requirements of the applications they are each running to ensure enough bandwidth for voice traffic over the WLAN. You will need to maximize performance by adding a sufficient number of wireless access points (APs) to each location where many users work. Keep in mind that since a WLAN is a radio frequency (RF) network, the physical environment will affect the coverage capabilities of each AP. Walls, glass partitions, and cubicle separators can affect the coverage area because these materials absorb signals. Take into account the physical characteristics of your organization and buildings and design your WLAN plan to meet these challenges. A physical survey before deployment will help you determine how many APs and switches you’ll need to meet coverage requirements. Keep in mind, however, that the more APs you add to a particular area will affect performance in terms of possible interference.
As mentioned earlier, you can meet wireless traffic needs by adding APs to any given area. However, there is also the risk of interference when too many APs are working too closely together. Be sure to plan carefully and run tests to ensure smooth call delivery so that crucial voice traffic is delivered. Load balancing is another scalability tool, which means traffic is load-balanced, or shared across APs, to ensure users are sent through the most available AP at any given time. The IEEE is working on a standard to allow wireless IP phones to discover all nearby APs available for service in order to utilize the most appropriate AP. Before that is available, some vendors are offering their own similar proprietary capabilities.
Quality of service
Delays for voice should not exceed 150 ms, and given that Wi-Fi is a contention protocol, when an access point is overloaded, voice quality will suffer. QoS is required for voice traffic whether it’s traversing a wired or a wireless network. In other words, you want QoS for your voice traffic over the air or over land so look for gear that offers over-the-air quality of service. Guaranteeing voice over other applications minimizes packet loss, delay and jitter that results in poor voice quality. The IEEE is working on a standard to address QoS for wireless networks, but in the meantime, the Wi-Fi Alliance has released Wireless MultiMedia (WMM) as a subset of these capabilities. Vendors are currently bringing WMM implementations to market now. WMM defines four priority levels to support varying kinds of traffic, including voice, video, best effort for data, and background traffic, in that order.
As a user walks from one office or location to another, he or she counts on roaming capabilities of the WLAN to keep the call connected. The underlying wireless infrastructure must seamlessly hand off the user to the next location and perform the necessary re-association and re-authentication with APs, while keeping calls free of interruption (this will allow a call to continue seamlessly across zones without being mistakenly dropped between zones). A security standard is under way to allow users to be pre-authenticated to neighboring APs before roaming, which will reduce the time it takes for a user’s call to move between APs, and in the meantime, some wireless equipment vendors are introducing their own versions of fast-roaming capabilities.
IEEE 802.1X authentication should be used to verify a user’s identity onto the network, which will ensure unauthorized guests are not allowed entrance to use the network or gain access to confidential corporate information. Laptops and handhelds can support 802.1X authentication, and you need to make sure your wireless IP phones, which have less computational capacity, are using less processor-intensive authentication methods like MAC address or username and password.
As discussed in Chapter 5, Session Initiation Protocol (SIP), a signaling protocol, is used for establishing a session in an IP network—from a simple two-way telephone call to a multi-media conference call session with many participants. The VoIP industry has recently adopted SIP, a RFC standard (RFC 3261) from the Internet Engineering Task Force (IETF), as the protocol of choice for signaling because of its ability to facilitate Internet applications by working with other protocols. Essentially, SIP establishes, manipulates and tears down sessions, and its main purpose is to help session originators deliver invitations to potential session participants wherever they may be. It uses URLs to address participants and SDP to convey session information and it’s easy to combine SIP with other applications, like Web browsers and messaging. SIP can find and invite call invitees wherever they are, and it facilitates multi-media calls with many participants who may join and leave at will.
It is possible to use traditional cell phones and they can become an extension of your IP telephony solution—this requires no new wireless network or SIP handsets. There are also many wireless IP telephony handset vendors out there, but they don’t all offer the same features. Start your search by looking at your needs first. Are you looking at wireless handsets for a manufacturing facility and therefore need a rugged handset with dust covers so they don’t get dirt inside the keys? Are you a healthcare facility and need to meet safety requirements so the handsets don’t interfere with hospital equipment? After you determine your general needs, next move on to what you would like to see the handsets offer. Would you like the handsets to be able to transfer calls? Would you like your employees to be able to conduct conference calls from the wireless handsets?
What’s your wish list on top of your needs list? These two things will bring you to a number of vendors’ solutions, and then the final question you need to ask is, will it work with your IP PBX vendor’s solution? Your choice will be very easy at this point—you’ll likely either have just one or two vendors left from your list.
The bottom line
You need to be prepared to establish wireless IP telephony because your users, customer service, and the bottom line will greatly benefit from mobile VoIP. You need to approach this the same way in which you approached the wired IP telephony system: First, look closely first at your basic wireless requirements in terms of handsets; next look at what features would be nice to have; and then look at what handsets will work with your infrastructure and your chosen IP telephony system. As you go through each of these steps, the number of solutions available to you will be reduced and you’ll be left with just a few options from which to choose. The next chapter will go deeper into Quality of Service plus cover Virtual LANs and MPLS.
Quality of service
As we drill down deeper into the details about your converged network, going over topics like handsets, security, and mobility software, you are most likely growing more comfortable with IP telephony. However, you probably still have some concerns, most notably, “How do I know my boss isn’t going to experience poor audio quality during calls?” or “How can I be certain that all of our average 200 calls are always going to get through, even on our busiest day, and when accounting is doing its weekly check run?”
The answer is Quality of Service, or QoS. This chapter will cover QoS in detail, as well as your options in terms of circuit transports, and then delve into the internal infrastructure and the entire process of applying QoS. QoS can be boiled down to three major steps: Identify. Classify. Prioritize.
Voice must be heard
Quality of Service is key when it comes to IP telephony implementations. Voice traffic must get prioritized so that it’s not delayed or discarded because of interference and congestion from other traffic. At the same time, you may also have other high priority applications, such as video or other business critical data, that also need higher than normal prioritization. But voice is always going to be high up on your priority scale.
You need to consider four things that can affect voice traffic:
- Latency (or packet delivery delay )
- Jitter (or the variation in time between packets)
- Packet loss (which can occur when too much traffic overflows buffers within the network causing packets to be dropped), and
- Burstiness (when your network undergoes bursts of packet drops due to jitter)
It’s bad business to have your voice traffic burdened by any of these effects. Distance alone on the WAN circuit can cause delay, as can lower-speed WAN circuits. Delays cause call participants to start interrupting each other because they believe the other person is finished speaking. Latency should not exceed 100 milliseconds (ms) one way for toll-quality voice and must not exceed 150 ms one way for acceptable quality voice. At 150 ms, delays are noticeable by the human ear, but callers can still carry on a normal, comfortable conversation.
Jitter can cause strange sound artifacts to contaminate the voice and users will complain of degraded voice quality. Jitter has many sources: network congestion, queuing methods used in routers and switches, or network routing policies such as traffic engineering or MPLS paths used by carriers.
If your phone conversations do not sound right and callers have to keep repeating themselves or have a less than satisfactory experience when they call, they’ll start looking for other ways to communicate with your employees, or worse, they’ll start looking to another company to serve their needs—one with which they can communicate more clearly. Your IP telephony system should sound better than your previous phone system— after all, that’s why you made the switch. It’s the only way you’ll ensure that you don’t lose business because of your technology change. IP telephony should increase—not decrease—your business and your bottom line.
WAN circuit transports
You’ve probably already chosen your circuit transport method, but the IP telephony exercise may have made you re-think everything. The following descriptions are not intended to list everything about your options but rather to ignite ideas about the differences so you’ll start questioning your options, which will help you make the best choice for your organization.
Leased lines are the most private way to go. They are also the easiest type of WAN circuits to configure guaranteed QoS. These circuits are direct point-to-point lines connecting your locations together. They can be used for data, including packetized VoIP, or Internet services.
Frame Relay circuits are more economical than private leased lines because the Telco providing the Frame service shares bandwidth among many subscribers. This can reduce your costs, especially for long distance lines, but commonly reduces your guaranteed bandwidth to less than your full circuit speed. Frame Relay can guarantee bandwidth and packet delivery only if you shape your outgoing traffic to match your committed information rate (CIR). Properly engineered, Frame Relay can provide a cost-effective means of transmitting IP telephony traffic and still guaranteeing QoS.
Asynchronous transfer mode (ATM)
All information sent over an ATM network is broken down into discrete packets. Unlike other packet technologies, ATM employs fixed-sized packets, each consistently at 53 bytes long. This means cell delay in ATM switches is predictable and manageable.
MPLS, like Frame Relay, is a label-switched system that can carry multiple network layer protocols. Similar to Frame Relay, MPLS sends information over a wide area network (WAN) in frames or packets. Each frame/packet is labeled and the network uses the label to decide the destination of the frame.
Speed your WAN
There are a number of products out there to speed your WAN. Known as “WAN optimization” products, they accelerate applications by eliminating redundant transmissions, staging data in local caches, compressing and prioritizing data, and streamlining chatty protocols. Other tools perform rate limiting to control the rate of traffic being sent to your network while more critical traffic, such as voice, is being transmitted, for instance. Rate limiting is performed by policing (discarding excess packets), queuing (delaying packets in transit) and/or controlling congestion (manipulating the protocol’s congestion mechanism).
Be sure you’ve covered all your bases with your service provider and created a service level agreement (SLA) that you are comfortable will guarantee you acceptable service delivery. If someone downloads a huge set of files from the Internet that bogs down the WAN circuit, is that going to cause a dropped call for your CEO or is your service provider going to have the right tools in place to make sure the call stays up and the download takes a back seat to the voice call?
A virtual private network (VPN) is a private network used by an organization or in many cases by a company and its partners or associates, to communicate or coordinate confidentially over a non-private network. VPN traffic can be carried over a public networking infrastructure such as the Internet. Internet-based VPNs offer the least amount of administrative control to regulate and guarantee QoS.
The needs of high-quality voice
Remember that you need to minimize latency, jitter and packet loss and ensure enough bandwidth so that you deliver high-quality voice. In order to do this, you must have complete administrative control over the equipment and the circuits, end-to-end, as well as all the tools necessary to ensure your system remains up and running smoothly one hundred percent of the time. This is often impossible based on budgets and equipment inventories. An alternative is to compromise in an area that allows you to save money while giving up only so much control as to still deliver high enough quality voice where degradation is barely recognizable or where it is entirely tolerable. In all LAN/WAN environments, there will be packet congestion—it’s inevitable. The key is to guarantee that VoIP packets are prioritized so that they are able to get through during those times of congestion, otherwise your QoS plan has failed.
Your spectrum of options runs from leased lines combined with feature-rich switches and routers at one end of the spectrum, to Internet-based VPNs using consumer-grade WAN circuits (DSL, Cable-modems) from separate providers with no SLA.
With the first option, you have complete administrative control over all points of congestion and have the configuration tools and features to easily identify, classify and prioritize your VoIP traffic. This is the optimal choice if you have the budget for it. Managed routers with features that you cannot control and circuits that you do not have administration control over can be less effective for your network and often require more labor to ensure configurations are correct and guarantees are being followed by the managed service provider. You have less power in terms of making forwarding decisions and changes on the fly. At the other end of the spectrum, you have cast off all control over every components, circuit and congestion point and have thrown your VoIP packets into the Internet with simply the hope that they get there, but effectively powerless to help them arrive safely and on time.
Latency (also known as delay) is the time that it takes a packet to make its way through the network to its destination (or the time it takes the speaker’s voice to reach the listener’s ear). Actually, some latency is inherent and constant due to distance and the number of devices in the path. As mentioned, large latency values can cause hesitations and, therefore, call participants interrupting one another. There can be a number of factors that contribute to latency, such as propagation delays (the time it takes an electrical signal to travel the length of a conductor), queuing delays, packet forwarding delays, etc. Again, end-to-end latency should be less than 150 ms for toll quality phone calls. Here are a few suggestions for mitigating the impact of latency contributors:
- The faster the media, the less time it takes to serialize the digital data onto the physical links, and the lower the overall latency. The impact on latency depends somewhat on the link technology used and its access method. For example, it takes 125 microseconds to place one byte on a 64Kb circuit. Placing the same byte on an OC-3/STM-1 circuit takes 0.05 microseconds.
- Although some delay is unavoidable regardless of the bandwidth used, keeping the number of intervening links small and using high bandwidth interfaces reduces the overall latency.
- The packet forwarding delay is determined by the time it takes a router, switch, firewall or other network device to buffer a packet and make the forwarding decision. Among the forwarding considerations are which interface to forward the packet to and whether to drop or forward the packet against an Access Control List (ACL) or security policy. Packet forwarding delay varies depending on the function and architecture of the networking device. If a packet must be further buffered as a part of its processing, greater latency is incurred. (Source: VoIP 101, Juniper Networks.)
Jitter’s impact on voice quality
Jitter, a variable delay, is the time difference between when a packet is expected to arrive to when it actually arrives. In other words, given a constant packet transmission rate of every 20 ms, new packets would be expected to arrive at the destination exactly every 20 ms. Unfortunately, as Figure 8.2 shows, this is not always the case. In Figure 8.2, packet one (P1) and packet three (P3) arrive when expected, but packet two (P2) arrives 12 ms later than expected and packet four (P4) arrives 5 ms late.
Jitter is caused by congestion or other factors. Most media gateways have play-out buffers that buffer a packet stream, so that the reconstructed voice stream is not affected. Play-out buffers can minimize the effects of jitter, but cannot eliminate severe jitter. Although some amount of jitter is to be expected, severe jitter can cause voice quality issues because the media gateway might discard packets arriving out of order. In this condition, the media gateway could starve its play-out buffer and cause gaps in the reconstructed waveform. (Source: VoIP 101, Juniper Networks.)
Tolerating packet loss
Packet loss is often unavoidable and can occur for a number of reasons, such as the case of a router or switch overflowing, and in many instances, applications can tolerate packet loss (as in the case of a noncritical file transfer). However, most real-time applications are less tolerant of packet loss. Although packet loss is not desirable, some voice packet loss can be tolerated as long as the loss is spread out over a large amount of users. If the amount of packet loss is very small in comparison to the users and over a large amount of time, then it can be acceptable.
When it comes to applying Quality of Service onto your enterprise network, it’s a matter of identifying, classifying and then properly prioritizing data and voice packets. Your first step is determining the method by which you will identify high-priority packets. There are a number of options available within an enterprise network to identify and mark which packets are high priority. These methods include VLANs, Differentiated Services Code Points (DSCP), also called DiffServ, Type of Service (ToS) bits, IP Precedence, 802.1p markings, Layer-3 IP-address, Layer-4 source & destination ports, etc. Once you’ve determined which method to use throughout your corporate network, you will use this method to identify, and possibly re-mark, each high priority packet. (Keep in mind, your IP telephony vendor may mark them with one method, and you may choose to re-mark the packets with a different identification method.)
Once each high-priority packet has been marked with your corporate standard (for instance, DiffServ), then at egress (as the packet leaves a piece of networking gear such as an Ethernet switch or router), it needs to be prioritized above other packets. Keep in mind there are different levels of priority as well as different queuing methods, so if your organization is a hospital, you will likely have healthcare applications, such as patient records and networked images transfer applications, assigned a higher priority along with voice traffic.
The following is a list, although not exhaustive, of identification methods you may choose to utilize for prioritizing voice traffic. Again, your IP telephony vendor may choose one method and you may choose another for your corporate standard, in which case you will be re-marking each packet with your method choice.
DiffServ or ToS
Layer 3 QoS using DiffServ or Type of Service (ToS) bits is a system of identifying IP packets by assigning values within the layer 3 IP header. Once identified, traffic can be classified into groups so that QoS policies can be applied. For example, maybe Web access needs to be reasonably responsive but acceptable e-mail response time can range from seconds to minutes. On the other hand, voice traffic (IP telephony) and IP videoconferencing require a much higher level of priority. The type of end-to-end QoS you choose to implement will depend on what type your routers and IP telephony solution support.
DiffServ and ToS add state information to each packet—allowing the network equipment to identify different service flows and direct queuing and forwarding treatment appropriate to the service requirements. This enables routers to identify voice packets and mark them for higher priority treatment over less sensitive packets. With DiffServ or ToS, each router on the network is configured to differentiate traffic based on its class and each traffic class can be managed differently, insuring preferential treatment for higher-priority traffic on the network.
802.1p is a specification that gives Layer 2 switches the ability to identify and prioritize traffic. It works at the media access control (MAC) framing layer (Layer 2) of the OSI model. Eight classes are defined by 802.1p, which uses the priority fields within the packet’s VLAN header to signal the switch of the priority-handling requirements.
A virtual LAN, known commonly as a VLAN, is a method of creating logically independent networks within one physical network. A few or many VLANs can coexist within such a network. For instance, a small 50person grocery store can have 10 VLANs dedicated to different departments of the store and one VLAN for information technology. A hospital could literally have hundreds of VLANs to segregate different staff members, doctors’ groups, departments, and labs. Administratively segregating and separating people and departments helps to reduce traffic on each VLAN so that each segment is performing optimally, aids in ensuring confidential information is accessed only by authorized personnel, and ensures that latency and bandwidth-sensitive traffic, like voice, is given priority. Often voice traffic is given its own VLAN (or multiple VLANs).
Once you’ve decided how you’re going to tag your high-priority packets, next you have to determine your prioritization method. Here are just a few.
Weighted fair queuing
Weighted Fair Queuing (WFQ) allows traffic flows to share link capacity but provides prioritization for small, time-sensitive traffic flows. The advantage is that a large flow will not clog the pipe or create lengthy delays for other smaller flows. WFQ is used in routers and switches that forward packets from a buffer that works as a queuing system where the packets are stored temporarily. Packets are essentially waiting in queues in buffer space, while WFQ estimates which packet flow will be “fastest” (the one with the minimum number of packets) and transmits those smaller, time sensitive packets ahead of the larger, delay-tolerant packets.
Priority Queuing supports multiple fixed-length queues from high to low, servicing the highest queue first, then the next-lowest priority and so on. If a lower-priority queue is being serviced and a packet enters a higher queue, that queue is serviced immediately. While good for important traffic, it can lead to queue starvation.
Custom Queuing is designed for environments that need to guarantee a minimal level of service to all protocols. It allows a customer to reserve a percentage of bandwidth for specified protocols. Customers can define multiple output queues for normal data and additional queues for system messages such as LAN keep alive messages. Custom Queuing can guarantee that mission-critical data is always assigned a certain percentage of the bandwidth, but also assures predictable throughput for other traffic. (Source: Custom Queuing and Priority Output Queuing, Cisco)
Class-based weight fair queuing (CBWFQ)
Weighted Fair Queuing classifies traffic into different flows based on layer 3 and layer 4 information, such as IP addresses and TCP ports. However, WFQ has some limitations— it’s not scalable as traffic increases, and native WFQ is not available on all high-speed interfaces. WFQ also doesn’t provide as much granular control as is often needed. CBWFQ provides a solution to these limitations. CBWFQ gives an administrator more control over what types of traffic classes are assigned to each queue and what unique prioritization methods each queue should be assigned including bandwidth, priority, queue size, reserved bandwidth, etc. The bandwidth you assign to a class is used to calculate the “weight” of that class. The weight of each packet that matches the class criteria is also calculated from this. WFQ is applied to the classes (which can include several flows) rather than the flows themselves. (Source: Understanding Class Based Weighted Fair Queuing on ATM, Cisco)
The bottom line
Quality of service is a must when it comes to real-time applications like IP telephony. Applying QoS is a matter of identifying, classifying (or marking), and then prioritizing voice packets. This chapter has outlined how to go from the outside (service provider) to the inside (infrastructure), from choosing the circuit transport to choosing prioritization methods. There are many options that can suit your needs and it will be a matter of discussing the choices with your service provider, integrator and/or colleagues. The next chapter will cover other options that are available to you from various service providers, including some new options being offered.
Outsourced IP telephony options
Up to now, we’ve covered in-house IP telephony systems. Because you have full control over systems that you own outright, you will find they are the most flexible, reliable, and scalable. With fully-owned systems, you’re also able to make the changes as soon as you need to make them—no waiting for a provider’s schedule to free up, no fees to be paid, and no miscommunication risks. However, hosted IP telephony is starting to gain some popularity. Savings are attracting businesses to hosted IP telephony solutions, especially small and medium-size businesses (SMBs) and those with multiple sites and organizations with highly mobile workforces.
A rising interest in cloud or hosted services is encouraging market growth for hosted unified communications solutions. The global hosted UC market is expected to grow at a rate of nearly 21 percent per year between now and 2015.
One of the major drivers for companies turning to hosted services is the need for centralized management. Enterprises are finding hosted UC a smart solution, as it offers a wide variety features that can be scaled to the businesses’ needs. These solutions are also managed off site by the service provider, freeing up IT personnel to focus their attention on other areas.
Hosted UC services can give companies a competitive edge by reducing operating costs, increasing flexibility and boosting employee productivity and efficiency.
Companies should look for scalability, flexibility, cost, performance management capabilities and network access requirements when choosing a UC service vendor.
Mitel Sky hosted VoIP and cloud unified communications services are available for organizations seeking a secure, managed business communications solution that requires no capital investment.
Its hosted cloud based telephone systems, hosted VoIP PBX, hosted UC, and hosted contact centers services give customers the flexibility to deploy the communications solution.
Outsourced IP benefits
In addition to the costs saved when organizations benefit from fixed rate bundles for local, national and international calls, outsourced IP offers improved effectiveness with features like voice mail and unified messaging. Outsourced IP telephony also offers the flexibility enjoyed by in-house IP telephony, in that employees can log in to any phone make and receive calls as if it were his or her desk phone. Voice mail can be retrieved remotely from any phone, including cell phones, and it can be retrieved via e-mail. Costs are more predictable because of price bundling, and quite often the system is more reliable because call re-routing for business continuity and disaster recovery is automatic. For a monthly fee, the hassle of managing your own telephony solution can be offloaded and a provider deals with the headaches plus the monitoring and management of your IP telephony system, in line with the service level agreements you’ve discussed.
Now SMBs have access to a cost-effective, feature-rich alternative to on-premises IP PBXs and traditional analog phone systems. Hosted IP offered by service providers specifically targeting SMBs offer modern features, scalability, and ease-of-use designed to give small and mid-sized businesses the competitive edge they’ve been looking for when it comes to their telephony systems.
You’ve got choices
Service providers understand the importance of being a one-stop shop for customers, providing data, voice and video services to customers, which is why they’re building tremendous multi-service networks to support it all. Customers now have all kinds of choices—they’re no longer locked into voice services only from their incumbent local exchange carrier (ILEC). The competitive landscape has changed and competing service providers (SPs) offer multiple services over a converged network. Companies can now look to a variety of sources for IP telephony services, including ILECs, competitive LECs (CLECs), Internet SPs (ISPs), and Value-Added Resellers (VARs). The competition has opened up the opportunity to all kinds of business, large and small, and provided competitive pricing and advanced capabilities to make IP telephony the right choice for many companies.
IP telephony features offered by service providers
- 3 or 4-digit dialing
- Call waiting
- Hunt Groups
- 3-way calling
- Caller ID
- International calling
- Automatic call distribution (ACD)
- Conference bridge
- Music on hold
- Authorization codes
- Direct inward dialing (DID)
- Operator console
- Auto attendant
- Distinctive ring tones
- Voice mail
- Call forwarding
- E911 compliance
- Web-based management portal
- Call park/call pickup
- Follow me
Managed IP telephony
In the managed IP telephony model, the company owns the IP PBX, either on-site or in the provider network, while the carrier provides oversight and maintenance on it and offers bundled services (caller ID, auto-attendant, call redirect, and other voice-related applications). The service provider also supplies the customer premise equipment (CPE) for packetizing voice before it enters the wide area, and also includes Service Level Agreements (SLAs) to cover Quality of Service (QoS) and support. Managed IP telephony is a great option for businesses that want to try out IP telephony but aren’t ready to manage the system.
According to Laurie Shook, Verizon Business’ director of managed IP telephony, when it comes to managed IP telephony, companies should:
- Evaluate service provider capabilities in terms of breadth of services and flexibility of offerings.
- Ensure the vendor is financially stable and committed to the business over the long haul.
- Determine whether resources are available when and where they’re required.
- Look for a service provider that will build upon the existing hardware and software investment.
- Identify the scope and scale of service provider responsibility.
- Tour the company’s network management facility and meet the people who will monitor the network.
- Ask about employee and site certifications.
- Select a service provider with built-in system redundancy.
- Obtain fully documented service resolution procedures.
- Consider vendors that are committed to continued investment in network operations and systems integration.
(Source: SearchVoIP.com, http://searchvoip.techtarget.com/originalContent/0,289142,sid66_gci1241254,00.html)
Hosted IP telephony
Hosted IP PBX delivers IP telephony services to subscribers with cost reductions and improved business processes and customers do not need to be tied to the switch. The architecture is similar to a Centrex system or KTS (Key Telephone System), except a service provider rather than a local phone company provides switching along with the gateways to the PSTN. It is fully outsourced, and the customer utilizes the broadband IP network for voice and data without having to own or manage the switch. The only CPE (Customer Premise Equipment) necessary for IP telephony are the phones or converters if analog phones are used.
With hosted IP, all IP telephony components, from media gateways and switches to application servers, are located at the service provider location (or data center). This model is different from a managed IP PBX model, where the equipment can reside either on the customer premises or in the service provider data center. In the hosted solution, the equipment supports many customers, whereas in the managed IP scenario, the IP PBX is dedicated to the use of one customer (not shared across customers).
IP telephony access service
Because many businesses already have their own PBX/IP PBX equipment and are looking to capitalize on that investment and keep network ownership costs down, IP telephony access service can help them do that. IP telephony access also serves as an introduction to converged network services. The service provider offers its customers whatever IP telephony capabilities it has in the data center— typically an abbreviated dialing plan, IP telephony VPN, hosted voice mail, and possibly a few others—all bundled together. The customer owns the PBX system so the service provider offers IP telephony through the same loop it’s already supplying to the customer (whether T-1 for large customers or broadband for smaller customers).
How to evaluate VoIP service providers
As more companies discover the benefits of cloud-based phone solutions, more phone service providers have entered the market. But not all business VoIP providers are created equally. In order to find the best hosted PBX provider for your business, here are questions to ask to get answers in a few key areas: technology, service and innovation.
Technology questions to ask VoIP service providers
- Describe your solution’s network infrastructure.
- Do you have redundant equipment configured?
- What are your recovery expectations for a major natural or man-made disaster?
- Did you develop your own soft-switch?
- Do you license any core technology from third parties?
Service related questions to ask VoIP service providers
- Do you have customers like me? Ask them to identify companies of similar, size, industry and complexity. Are there case studies, customer testimonials or videos that you can review?
- Do you integrate with my business applications, such as salesforce.com?
- What are the initial, monthly and ongoing maintenance and support costs?
- Will I have to pay for upgrades to the system?
- How long is the contract?
- What are the details of your phone system implementation process?
- Will someone be provided to help me with the implementation or am I on my own?
- Who will support my users if they have questions or problems?
- Do you have a service level agreement in place?
- Do you provide dedicated circuits to ensure high quality of service?
- Do you make your performance and reliability statistics public?
How innovative are the top VoIP providers?
- What is your product roadmap for the next year?
- How are development priorities decided?
- Are customers consulted as new features are developed? If so,what is the mechanism for doing that?
- What is the size of your engineering and software development organizations?
- What can you tell me about your financials? How do I know you will be around to support me for the next five to seven years?
The bottom line
Outsourcing your IP telephony services may not be a fit if your organization is a midsized enterprise and you want full control of your IP telephony system, including control of your equipment and the ability to make changes to it on the fly, including moves, adds and changes (MACs). However, if your organization is a geographically scattered SMB struggling to look to the outside world like a larger unified entity, then a hosted IP solution may be the way to go until a time in the future when you’re ready to bring an IP telephony system in-house. With a hosted IP telephony system, you can streamline communications, boost productivity and customer responsiveness, and reduce your communications costs. You can improve your bottom line and increase your customer satisfaction.
The next chapter will cover system management, from configuration and troubleshooting and MACs to sophisticated reporting tools that can help you improve your overall business planning.
Ease-of-use and management
In order to lower the total cost of ownership (TCO) of anything, you must address the human resources required to operate, manage, maintain, and support it. Ease of management is especially vital to TCO when choosing an IP telephony system. The network convergence you’ve already conducted allows you to stop maintaining parallel networks for voice and data and instead have a single, converged infrastructure that leverages investments and streamlines administration and management. Your chosen IP telephony solution should take this even further, simplifying the end-to-end management of your IP telephony system, from device to network monitoring and analysis. An IP telephony solution that’s truly easy to implement and manage can pay for itself very quickly–often in a few months.
Ease-of-implementation and administration
You probably have similar concerns about IP telephony management as many of your peers. Your IP telephony solution should be easy to implement as well as easy to manage. It’s important that you are able to easily add new locations, branches, or remote offices if expansion is in your business plan through regular growth or acquisition. Generally, voice platforms that have been designed from the ground up to be IP telephony platforms have management interfaces that are easy to use and make changes quick and easy. Generally, administrators with basic networking skills can easily be brought up to speed on purpose-built IP telephony solutions. It’s often a different story with analog phone systems that have been jury-rigged to “look and feel like” IP telephony.
In general, it is very simple to add a new user to most IP telephony systems. It’s usually just a matter of typing the user’s name into an easy to use form and editing or accepting various default fields. With these simple steps, the user is given an extension, direct inward dial (DID) number, if the IP telephony vendor offers the feature, and a voice mail box. Depending on the IP telephony solution, various features are updated, such as automated attendant, dial-by-name and number and online directories.
As shown in Chapter 1, Aberdeen’s TCO study showed that respondents preferred the ease of use and simplicity offered in an IP-based telephony system such as Mitel’s.
The study showed that for recurring costs, the largest benefit was staffing, and Mitel users required 88 percent less Full-Time Equivalent (FTE) staff to manage their system on an ongoing basis than others.
“This is likely attributable to the reduced complexity and increased ease-of-use of the Mitel system,” Aberdeen said.
In the software support and maintenance category, Mitel also showed a significant advantage in cost savings. “Again, a well-designed and easy-to-use system typically requires less support and ongoing maintenance,” Aberdeen said.
Easy toll bypass set-up
Despite the fact that long distance rates have dropped dramatically in recent years, there are still savings to be had in toll bypass. However, the process of setting up the network call routing for toll bypass must be easy to follow and simple to carry out in order for the savings to outweigh the management overhead. In IP telephony implementations built on legacy voice and data platforms, automatic peer-to-peer exchange of newly entered routing information is not automatic. This routing information must be entered into routing tables—and possibly even into each individual PBX or physical router. Small, localized groups of such devices may automatically update each other, but the information doesn’t get pushed out to every switch in the network.
Given the capabilities of the IP environment, network managers shouldn’t have to define specific routing behaviors for each location, and users shouldn’t have to remember which area codes qualify for toll bypass. Long-distance calls to outside numbers that fall within the local dialing radius of one of your company’s sites should automatically get routed over your IP backbone, a process that should occur transparently to the caller. With a distributed system, you should have access to an intuitive graphical interface to replace those routing tables, and toll bypass routing will take place intelligently and automatically.
Simplicity for end users
In many cases, most companies that have already deployed IP telephony have not even scratched the surface of utilizing all of the capabilities, usually because the features are too hard to implement. If something takes time away from your job as IT manager to train end users and re-train them on phone features and other system features (such as logging in from another phone, or using the follow-me feature), these training costs (time and budget) outweigh the benefits. You want a system that’s “self-service” for end users, and one that offers a very intuitive set of features. Setting up a conference call should be easy for anybody to handle—you should not be brought in to set up something every user should be able to utilize on the fly. Otherwise, the benefit is lost in the complexity and cost.
The best IP telephony systems have a single intuitive user interface that enables nontechnical users to help themselves to such functions as call management, setting up conference calls, logging in to work from a remote location, and managing their own integrated desktop communications. When IP telephony is implemented properly, end users should be taking self-service to the highest level and IT managers or administrators should only need to step in when new users must be added or permanent changes have to be made. When this self-service model is followed, burden is lifted from the IT or network staff, who can then spend their time on other revenue-producing or business improvement tasks.
Benefits to the contact center
In addition to saving time and money when it comes to IT staff budgets, IP telephony should reduce contact center operating costs. The cost savings associated with an IP contact center can quicken the return on investment (ROI). First there are the savings associated with having the contact center built on IP telephony because you eliminate T1 charges between sites where contact center personnel reside. Management is also centralized for the contact center, even though your agents may sit in multiple locations around the globe. You can manage the entire statewide, countrywide or global contact center from your own location.
When it comes to the contact center for overall reporting, your IP telephony system should offer as high-level and in-depth information on call handling as you need to maximize your organization’s staff. Contact center reporting applications should enable managers and supervisors to generate historical statistical reports to assist in evaluating past activities and planning future actions. Some vendors offer predefined reports that can be generated as they’re described, or you can alter the fields and create customized reports for various departments and/or executive scrutiny. The generation of these reports should be as easy as the entire system is to use via drag-and-drop commands from a centralized desktop.
Several factors contribute to the ability of contacts centers to lower costs. They are purpose-built and designed for ease along with a number of productivity features, including a unified, intuitive desktop interface used company-wide: IVR, intelligent routing, outbound, and media handling.
Through complete unified communications integration, contact centers can now achieve single call resolution with functionality that lets agents communicate both externally and internally effectively using voice, Web chat, email, instant messaging and video. Agents and supervisors can also take advantage of the latest telecommuting features while still leveraging powerful dashboard, reporting and unified management capability.
Built-In system monitoring
Your IP telephony system should provide you with a comprehensive monitoring application that collects network utilization and error information, as well as server and device performance statistics, to identify potential problems that may impact your IP telephony call quality. This can help you quickly identify issues and proactively make changes before the issues become problems. For instance, you should be able to view current network as well as historical utilization of any monitored link, and historical packet loss information so you can make infrastructure tweaks. Whenever new devices are added, the monitoring application should detect these changes and make automatic configuration updates, and it should scale easily to provide full functionality with minimal maintenance, no matter how large your network grows. Network analysis is a must-have for your IP telephony system so that you can proactively maintain an optimal network and reactively make changes so that call quality does not degrade.
A new architecture for UC
Zeus Kerravala, founder and principal of ZK Research, writes that VoIP and UC are not like traditional voice services, so the underlying architecture that supports them needs to change. IP-based applications tend to be centralized and then distributed over an organization’s WAN. Kerravala cites the example of a Web-based application. The Web runs at Layer 3 of the OSI stack, or the IP layer, meaning it has the ability to traverse the entire length of a network as long as it’s properly configured.
“When an organization deploys a Web application or any other IP-based corporate application, the servers are centrally located in the company’s data center and the endpoints (laptops, PCs, mobile phones, etc.) make a request back to those servers to fetch the required information. It doesn’t matter where in the world the user or the servers are, the communications between the client and servers work because of IP,” Kerravala wrote in a 2011 article entitled, “Accelerating Unified Communications With an Enterprise wide Architecture” published by The Yankee Group.
“The new architecture needed for UC would follow the same deployment model of today’s Web- and IP-based applications, using a centralized architecture based on standards-based, loosely coupled components,” Kerravala added. “In this architecture, telephony and other UC services would be centrally deployed services that could be distributed to remote locations over the WAN, just like other corporate applications. This type of centralized architecture is significantly simpler than traditional voice architectures and will scale with the organization. The architecture is characterized by:
- Communications applications deployed centrally in corporate data centers and distributed over the IP network
- Foundation based on industry standards such as SIP and XML
- Three-tiered architecture that removes the dependencies among the user devices, access points and applications
- Support for multivendor environments, and mixed legacy and new IP systems
- Ability to add new features incrementally without having to forklift upgrade
- Use of central SIP trunking that is shared across the enterprise, reducing the need for local trunking at separate locations
- Ability to securely bring more advanced SIP-based consumer, service provider and cloud services into the enterprise
“Using a more loosely coupled architecture lets service providers roll out new services to millions of subscribers globally,” Kerravala wrote. “Enterprises need to become service providers themselves and leverage similar principles and ideas to more easily serve their own people, wherever they are. Using the same SIP standards within the enterprise will also simplify the connection between enterprise communications and external service providers—starting with SIP trunking but expanding to rich media services in the future.
“This new UC architecture offers many benefits, primarily due to overall design simplification,” wrote Kerravala. “With a traditional architecture, connections need to be made between every location, creating a management burden. With a Web architecture, all of the data is sent back through a centrally managed infrastructure. This type of architecture delivers the real value of IP because it is truly manageable, another key to successful deployment and adoption of business-critical UC applications.”
The bottom line
Lowering TCO requires simplicity. Your IP telephony system should be as easy for employees to use as possible, requiring fewer interventions by and support calls to the IT department. Initial training can be expected, of course, but beyond an initial training session, users should be able to easily master features on their own with very little effort. It should also be easy to manage and help you make the most of your organizational resources. When voice platforms have been built from the ground up, this is generally what you get—a seamless system that is reliable, easy for users to learn and navigate on their own, and easy to manage.
Determining the right architecture (centralized or distributed) comes down to your organizational needs, resources, and business requirements. Distributed systems are a fit for organizations that have a lot of branch and remote offices and companies that grow rapidly and need to scale easily. The centralized architecture is geared to large enterprises supporting tens of thousands of users with large IT staff centrally located to support the system.
The next (and final) chapter will highlight some IP telephony implementations as they’ve been covered in the press. Chances are, you’ll find an example that closely resembles your own situation so you can make comparisons and learn from peers who have already implemented IP telephony.
The bottom line is that IP telephony can help your organization gain a competitive advantage, boost employee productivity, and enhance customer service. The real proof of these IP telephony benefits lies in the budget, resource and time savings, as well as the actual end user experience.
This chapter will highlight successful IP telephony implementations. The case studies will highlight different size organizations as well as companies within different vertical markets.
Wisconsin school district recommends Mitel for lowered TCO
Green Bay Area Public School District deployed Mitel solutions for significantly lowered TCO and collaborative, cost-effective communications among teachers, staff, and parents.
“In this economy, no one can afford to take a chance on big complicated systems, even from brand name vendors. We wanted a complete, straightforward, easy-to-use reliable solution, and Mitel demonstrated how it was possible—brilliantly simple became very evident.” - Allen Behnke, Director of Safety, Security and Telecom, Green Bay Area Public School District.
- With an antiquated voicemail system and burgeoning communications requirements for 40 locations, the Green Bay Area Public School District needed to take a comprehensive look at the districts telecommunications system rather than doing piece meal repairs and replacements
- Green Bay selected a Mitel UC Solution for intuitive, secure and brilliantly simple collaboration across its school district. The solution comprises: more than 2,800 Mitel IP Phones (models 230g and 115), Mitel Voice Switches (models 220T1A, 90, 50), Mitel Communicator with Professional and Operator Access, Mitel Converged Conferencing Bridge, and Mitel Emergency Notification Application.
- Achieved a lower total cost of ownership while providing superior communication service and collaborative capabilities, by reducing administrative, training and support costs.
- Reduced the number of phone lines by 80 percent and trimmed monthly service costs by 70 percent, while improving communications and security across 40 locations
AP+M propels productivity and savings with Mitel Mobility
A turbine engine parts distributor uses Mitel Mobility to simplify smartphone integration and capitalize on Mitel UC features and cost-saving call capabilities.
“Mitel helps cut international roaming and long-distance costs. We’ve already cut our phone bill by 25 percent.” Angelo Fraioli, VP and Controller, AP+M
- AP+M needed to replace a limited PBX system to improve communications with customers, simplify connectivity for its mobile workforce, and save money on long distance phone charges.
- The Mitel UC Solution includes: Mitel Mobility licenses, Mitel Unified Communications and a Salesforce.com contact center Adapter
- Mitel Mobility extends UC capabilities from the desktop to mobile devices to boost productivity for on-the-go employees, allowing AP+M to slash long-distance usage and costs by 25 percent.
- Mitel’s Salesforce.com contact center Adapter unifies call routing and manages all business communications from one application via the use of Mitel Softphone, helping AP+M to automate call logs and use screen pops to accelerate call completion.
- Mitel’s single-image architecture and intuitive collaboration tools afford each AP+M employee an individual extension and easy Find Me, Follow Me forwarding capabilities, eliminating the need for a switchboard and saving $45,000 annually.
Mitel’s open API and workgroups drive greater efficiencies for trucking company
A large transportation company gets behind the wheels of IP progress with a Mitel UC solution, for simpler, smarter and cost-effective business operations.
“Users initially resisted the new technology – that evaporated overnight. Mitel is so easy to use and manage. Now the company can’t imagine using anything else.” - Dan Lyddy, Vice President of Information Systems, DART Transit Company
- DART Transit Company needed a new communications system to support a unique call routing environment, custom applications and to provide visibility, flexibility and new functionality.
- DART implemented a Mitel Unified Communications (UC) solution, including Mitel IP Phones, Mitel T1 Voice Switches, Mitel Communicator, Mitel VPN Concentrator and the Mitel Converged Conferencing solution.
- Mitel supports more than 200 workgroups providing rapid response to customer calls, driver calls and safety matters.
- Mitel’s open APIs let DART write and integrate custom applications for optimizing business processes. Call wait times have been reduced from 20 minutes to rarely a queue.
- Costs have been reduced due to the elimination of many T1 lines, extra data line requirements, and the need for outsourcing phone maintenance.
- Mitel’s VPN solution has eliminated the need for POTS lines and improved performance, continuity and reliability from distributed locations.