VoIP technologies for businesses

VoIP technologies and phone services for your business

Voice over IP (VoIP technologies) has gone from alternative technology to the primary way that businesses and many homes implement voice communications. However, as VoIP evolves its features and capabilities finding the right solution for your needs can be complicated and expensive.

If your a small to midsized business (SMB) operator, finding the right VoIP solution for your particular situation can be one of the more complex business IT decisions you’ll face. A good rule of thumb: It’s best to start with the basics. Examine your company’s current phone solution and assess its capabilities, cost, and your overall satisfaction.

Then meet with your front-line business managers — the folks that actually use it to help generate revenue. Pull your IT people into that conversation. Find out exactly how they’re using the system, where the pain points are, and what the wish list might be. Don’t let IT do much talking, just have them listen and take careful notes.

With that data in hand, write yourself a handy Phone Plan. What you’ve got, how much it costs, what your employees’ overall satisfaction is. Follow that with a wish list of features you think you need right away and those you think you might need in the near future (1-5 years is a good rule of thumb). This is where IT can start talking, mapping their knowledge of current VoIP and UCaaS capabilities to the needs your employees have expressed.

Figure out how many employees you might have in 5 years so you have some idea of how much the new system will need to scale. Depending on your business, you might also want to talk to legal and see how any compliance or regulatory requirements might impact your selection process, too. Now you’re ready to start taking demos from prospective VoIP VoIP technologies providers.

If you’re a consumer looking for a home-oriented service rather than a business, then the process is similar, though much less complicated. Figure out what you want from your phone system and start comparing. Unfortunately, while there are still independent VoIP technologies vendors that have consumer-oriented offerings, most if not all of them are much more focused on business sales. That’s because it’s hard to beat your local internet provider on cost.

Whether it’s Comcast, Verizon FiOS, or any of the many other home internet service providers out there, the vast majority offer some variation on the “triple play” package: Internet, cable TV, and phone. Not only is this convenient, but it’s also usually equipped with decent features and comes at a very nice price.

To opt out of that in order to bolt on another third-party service probably means you’ve got some specific needs that standard VoIP systems don’t provide. If you’re having difficulty figuring out what those might be, you can start by understanding what VoIP really is.

What Is VoIP?

The way in which voice signals are transmitted nowadays has evolved significantly since the original invention. In the beginning, the method involved a very simple point-to-point connection between two devices. Over time, the concept of switching was introduced, enabling the routing of calls to multiple devices by using an operator to physically “switch” a user’s phone and connect the call to an incoming line.

Early versions of automated call switching used electromechanical switches to connect outside lines to a specific handset. With the introduction of digital circuits came the ability to automate the switching functions, making it simpler for the user to make and receive calls.

The concept of the exchange came about as the number of telephone lines began to grow and the grouping of numbers by a central office or local exchange became necessary. This concept used the idea of a three-digit exchange number, which is still used today, as the first three numbers after the area code.

With a four-digit extension number, a single exchange could, theoretically, handle up to 10,000 numbers from 0000 to 9999. The same idea was used for large corporations with thousands of extensions in the form of a Private Branch Exchange (PBX).

VoIP is another evolutionary step in the transmission of voice over long distances. At the most basic level, it involves encoding voice into a digital form, which can then be transmitted over the internet. On the receiving end, the encoded signal must be decoded to let the recipient hear the sender’s voice. Other pieces of the puzzle include Session Initiation Protocol (SIP), which handles the set up of a VoIP call (see below).

Most VoIP technologies providers incorporate a cloud-based PBX as a part of their product. This functions in much the same way a traditional PBX works on-premises in that it connects calls destined for a specific endpoint to an incoming line. Another key function of a cloud-based PBX is to provide a Public Switched Telephone Network (PSTN) gateway to facilitate the connection of VoIP calls to a physical phone number.

What Is SIP?

SIP stands for Session Initiated Protocol, and you’ll bump into it a lot when evaluating solutions or testing your network for VoIP readiness. It’s important because, for the most part, it’s the de facto standard you’ll be using every time you have a VoIP conversation.

When VoIP first entered the scene, equipment manufacturers and software developers, especially for the largest enterprise players like Cisco or Nortel, developed their own proprietary protocol standards. They did this in part because it was easier, but also partly to keep customers “trapped” on their systems.

Once you were up and running on a proprietary VoIP protocol, switching providers meant a complete “rip and replace” where both hardware and software would have to be switched out. This not only drove up costs but also meant customers would lose their investment in the previous solution.

As VoIP technologies grew in popularity, small vendors building cheaper and more easily managed solutions began settling on SIP as the most oft-used protocol allowing easier communication between systems. Still, other VoIP protocols still exist, with perhaps the second-most popular protocol being H.232.

What makes SIP so popular is not only that it’s deep and flexible, but also because it was purpose-built to engage in multimedia (meaning not just audio but also video and even text) communications over TCP/IP networks. For VoIP calls, SIP can set up calls using a number of IP-related protocols, including the Stream Control Transmission Protocol (SCTP), the Transmission Control Protocol (TCP), and the User Datagram Protocol (UDP), among others.

But it can also handle other functions, including session setup (initiating a call at the target endpoint—the phone you’re calling), presence management (giving an indicator of whether a user is “available,” “away,” etc.), location management (target registration), call monitoring, and more. Despite all that capability, SIP is simple compared to other VoIP protocols primarily because it’s text-based and built on a simple request/response model, and is over very similar to both HTTP and SNMP. Yet, it’s still capable of handling the most complex operations of business-grade PBXes.

SIP is built to work on a peer-to-peer (meaning computer to computer) basis. In SIP-speak, the two points are called the “user-agent client” and the “user-agent server.” Remember that those are swappable, meaning that unlike other client-server protocols where the client is always the client and the server is always the server. In SIP, the endpoint making the call is the user-agent client initiating the traffic and endpoint receiving the call is the user-agent server receiving the call.

If the call order is reversed, the client-server nomenclature is reversed as well. That adds a lot of flexibility, but it also means every endpoint needs to be able to perform both roles—server and client.

There’s a long list of additional network elements to a full SIP solution, but two important ones are the proxy server and the gateway. The proxy server helps lighten the functional requirements of SIP endpoints. It also acts as both client and server, but it adds functionality around call routing and policy-based management. SIP gateways are the closest analogy to an old-style PBX network in that they can handle the routing and connectivity requirements for connecting SIP calls to other networks.

Typically, the advanced features of the VoIP technologies vendors we review here are largely based on the proprietary management technology they build into their proxy servers and gateways. By offering VoIP solutions where these (and more) elements of a SIP solution are hosted in the cloud, the providers reviewed here have more flexibility in building advanced features since they have more control over deployment and reliability.

VoIP technologies Basics

While understanding the basics of VoIP and SIP is important, setting one of these systems up will require some general network knowledge, too. For the best quality, you will need to meet a minimum upstream and downstream data throughput requirement.

In addition, you’ll also need to meet a minimum latency number (that is, the time between when a signal leaves a remote computer and when your system receives it), typically measured in milliseconds. It is possible to test your network connection to see if it will support a VoIP service. RingCentral offers this service from their website.

Home VoIP technologies are usually very simple to set up. You get a box or a special phone adapter unit from the service provider, and then you plug it into power, your internet router, and a standard telephone handset. An alternative that’s becoming less common is to purchase a SIP phone, which includes electronics enabling it to sidestep the need for the adapter.

You might have to adjust your network a little to make sure your call quality remains good, but usually, your VoIP provider can help with this step. If you’re an SMB, setup is more complicated, not only because the feature set is much deeper, but also because you’re talking about far more users, a wide variety of other network traffic that also has performance requirements, and business-grade security issues.

Business VoIP technologies

The business-focused entrants in this roundup, as you might expect, go much farther in additional capabilities, serving as PBXes. These let your business present a professional appearance to the telephoning world with features such as automated attendants, call recording, call routing, and conference call bridges.

Interactive Voice Response (IVR) systems replace the need for an operator to route calls to departments or individuals using dial-by-name. For customer support, the use of call queues—complete with hold music and wait times—help enhance the customer experience.

On the higher end of this space, hosted PBX providers such as RingCentral Office and Fonality Hosted PBX will generally require some on-premises hardware such as specific desk and cordless VoIP phones preconfigured to work with the hosted PBX service.

These phones connect to the provider over the internet and function exactly as you would imagine a business phone should, but the phone system running those phones are located in the cloud rather than the telco closet in the basement.

Self-service management and configuration of these systems generally occurs through a web-based portal and can include a long list of potential features. For SMBs, the most commonly important features you should be considering include:

Interactive Voice Response (IVR) systems, “Press 1 for accounting, press 2 for the White House…”

Call queuing, generally used in call centers, where systems like this distribute incoming calls to specific recipients based on what the caller wants, extensions dialed, or other criteria.
Hold music or audio, which should have not only a good list of options offered by the service but also the ability for you to upload custom music or audio.

Extension assignments, meaning an administrator on your side of the relationship should be able to assign internal extensions as desired. Number porting so you can use your current business phone number with the new service (important for folks who pay for 800, 888 or similar lines).

Call recording so you can use phone experiences for training, sales, and marketing intelligence purposes. Voicemail to email transcription so your employees can read or play their voicemail from wherever they receive email.

In some cases, providers will offer on-premises Public Switched Telephone Network (PSTN) connectivity through hardware that is connected to analog or digital phone lines from the local telco, and connected to the business network. This allows a business to continue to use local phone lines with their hosted PBX solution and may be of significant benefit to businesses that have a requirement to maintain local wired lines.

Furthermore, most (if not all) providers also offer smartphone integration with custom and third-party apps, like CRM systems that can extend the phone system beyond just basic voice communication. Such integrations can also allow users to transfer calls to and from their mobile phone, place and receive calls from their personal phone (that appear to be coming from the business), and interact with colleagues and customers via voice and text.

The Right Internet Connection

Most of these VoIP solutions will require stable and consistent internet connectivity at every location where the wired phones are to be used. In many cases, standard business-class internet service with suitable bandwidth will suffice, though the use of Quality of Service (QoS) configuration on a business-class internet router may be necessary to prioritize voice traffic over other internet traffic in order to maintain good call quality.

Some hosted PBX providers offer assistance with this type of configuration on existing customer hardware, assuming that hardware can support QoS configuration. Other providers will sell a specific piece of network hardware with the proper QoS configuration for the business to install that will ensure that call quality is prioritized.

Some providers, such as Citrix Grasshopper, offer a solution that doesn’t use VoIP at all. They are essentially just simple PBXes that consider existing phone lines to be extensions and route calls that way. For instance, you might have the main number that delivers callers to an IVR system and, when the caller dials an extension or selects a destination such as “Sales” or “Support,” the hosted PBX calls an existing landline or mobile number and connects the two calls.

The caller is unaware that they have connected to a completely different phone number, as the system looks and functions like an in-house PBX with call forwarding, transfer, hold music, IVR, and so forth.

The extensions in this case could be Plain Old Telephone System (POTS) lines, mobile phones, or even VoIP phones through a different provider. All the hosted PBX cares about is that, when a certain extension is selected, a call is placed to the phone number assigned to that extension.

This sounds basic, but it’s a tried-and-true technology that can make businesses of any size and budget look as if they are using enterprise-grade phone software—without the need to invest in heavy-duty PBX solutions or dedicated desktop phone hardware.

Optimize Your Network for VoIP technologies

In addition to making sure your internet service can handle your VoIP traffic, you also need to make sure your local area network (LAN) can handle it. What makes it tricky is that if you simply drop VoIP onto your network, that traffic will get processed the same as any other traffic running across your LAN, like your shared accounting application or those 20 gigabytes worth of files your assistant just stored in the cloud.

The problem there is that VoIP traffic is much more sensitive to network bumps and potholes than most general office traffic. That translates to the sound breaking up or cutting out entirely, difficulty connecting over Wi-Fi, or (worst case) dropped and lost calls. Fortunately, most of the providers reviewed here have an engineering staff that will contact you as part of your setup process to help your IT staffers test and optimize your network prior to deploying their solutions. That’s definitely something we recommend, but there are steps you can take now to prep your LAN for VoIP and make the deployment process that much easier.

Understand QoS: Quality of Service is the primary mechanism for keeping VoIP traffic flowing smoothly. It prioritizes specific traffic on your LAN ensuring that certain streams (in this case the VoIP traffic) get priority and always have a certain percentage of the overall pipe available to them.

Codecs: Have your IT staff familiarize themselves with the codecs being used by the VoIP system you’re considering purchasing. Usually, you’ll have options, meaning multiple codecs to choose from. Testing these out during your evaluation period will let you pick the best codec for your environment.
Network Monitoring Tools:

If you’ve got a LAN with more than 10 users, then you’ve likely hired at least one IT staffer and that person is using some kind of tool to monitor that network, including not just the health of connected devices, but also the kinds of traffic flowing over it. Prior to deploying your VoIP technologies system, it’s a good idea to make sure the tools currently being employed can also effectively monitor and manage VoIP traffic using common management protocols like SNMP or outright packet sniffing.

Once you’ve engaged with a VoIP technologies provider, their engineers will help you determine the overall service grade of your network (look at that as your network’s basic “VoIP readiness factor”) and how to tweak their service to run effectively over your infrastructure.

If it turns out you need to upgrade some of your local networking infrastructure, this process will determine that, too, so wait for it to complete before dropping any dollars on new routers or switches. Contact Musato Technologies to learn more about how VoIP technologies can assist and transform your business.

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