Author: Casey Houser
How Your VoIP Service Reaches 99.999% Uptime
Posted on November 20, 2018 by Casey Houser
At VirtualPBX, we take a lot of pride in our business phone plan features, including our VoIP service uptime, which we advertise as 99.999 percent.
That last digit might seem showy. We don’t mean to brag, though. The final .009 addresses an important issue: we deliver phone system reliability that not every service provider can match.
Our claim of that last digit is anything but simple. We operate six servers across the globe, work with multiple phone carriers, and protect our networks with enterprise-quality firewalls and backups to assure that your phone system is reliable.
99.9 vs 99.99 vs 99.999 Percent
It bears repeating that the figure — 99.999 percent — isn’t meant to be flashy. When you round to that many places past the decimal, you actually see some significant effects.
There are 525,960 minutes in a year. When you break those minutes down by the whole percent, you end up with 5259.6 minutes in each percent. In other words: 525,960 / 100 = 5259.6
This means that, if we only offered 99 percent uptime, you could see outages in your phone system, on average, of 5,259 minutes a year. That’s almost four whole days! It’s clearly unacceptable.
You can extend those calculations to the tens, hundreds, and thousands place by dividing the number of days, respectively, by 1,000, 10,000, and 100,000.
| % Uptime | Downtime per year (minutes) |
|---|---|
| 99% Uptime | 5259 |
| 99.9% Uptime | 525 |
| 99.99% Uptime | 52 |
| 99.999% Uptime | 5 |
What we strive for at VirtualPBX is only five minutes per year of downtime. This industry gold standard is extremely difficult to reach, but it’s achievable through a number of efforts.
Global Servers
No VoIP service provider is equipped to properly handle enterprise clients without multiple servers to their name.
VirtualPBX’s global network of servers makes it possible for us to work with clients worldwide. They also assist with our reliability because we can lean on them as fallbacks for when a local outage occurs.
If, for instance, one of our U.S.-based servers experienced a power outage, we could immediately route traffic from that location to another inside the U.S.
Our clients’ calls would still connect because the second server picked up the slack.
Firewalls and Backups
In our servers, we use firewalls to protect malware from getting in, and we use backups to make sure your company information always remains accessible.
Our firewalls provide double-duty – like much of what you see in this article. They help keep the information in our servers safe by protecting them from internet malware and keeping bad actors at bay. They are essential for keeping yours and our data safe so the whole system can function as expected.
Similarly, our backups reach across multiple physical locations to provide a redundancy for both VirtualPBX processes and client information. Our global network gives us the ability to move important data from one secure lockbox to another. Everyone in our ecosystem remains protected and accessible.
Multiple Phone Carriers
As VoIP service provider, we connect calls through internet connections. We work with various phone carriers – think AT&T or Bandwidth – to connect calls to other VoIP devices, smartphones, and landlines.
In fact, the process of connecting calls can get complex, as we discuss in our feature about how VoIP calls use the PSTN.
The way we’re able to hand calls between users is by working with carriers. Our portfolio of relationships with these companies means we can reach phones across the globe, and it also means we can increase our uptime.
We aren’t limited to handing off calls to a single carrier. Therefore, if one of our carriers experiences a problem, we can look to others to help route calls to the proper destinations.
You can read even more about the structure of the public telephone network in our comprehensive guide, What is VoIP?.
Does Your VoIP Service Measure Up?
We maintain and bolster our 99.999 percent reliability by doing everything we can to protect your data. We secure our relationships with carriers and work hard to continue our position as a leading VoIP service provider.
Does your current service provider measure up? If your service isn’t what you expect, it could be time to switch to a better business phone plan.
With Robocalls On the Rise, FCC Issues Ultimatum to Telecoms
Posted on November 15, 2018 by Casey Houser
If you have received an automated call to your home phone or smartphone recently, you’re not alone. The Federal Communications Commission (FCC) estimates that, if nothing is done to combat the problem, robocalls could represent nearly half of all calls in 2019.
The The Washington Post reports that this increase in robocalls is part of a multi-year trend where spam has jumped from 3.7 percent of total calls in 2017 to 29 percent this year. This dramatic increase has finally forced the FCC into action. An FCC press release this past week says the federal organization “will do everything we can to catch and stop spammers” and that it’s urging VoIP providers to “implement tools to speed the traceback process” to catch spammers in the act.
Furthermore, the FCC issued letters to 14 prominent telecommunications companies demanding that they implement the SHAKEN/STIR method of call protection. The Commission’s rhetoric includes the promise that it will “take action” if the telecoms do not fall in line.
What is SHAKEN/STIR?
The Signature-based Handling of Asserted Information Using toKENs (SHAKEN) specification and Secure Telephone Identity Revisited (STIR) protocol provide methods for caller identification. They appear to be the best chance the telecommunications industry has in defending the public against robocalls.
You can think of these protocols as the equivalent to websites that use the https:// communications protocol. In short, https:// sites differ from ordinary http:// sites because they use a method of encryption called Transport Layer Security (TLS). TLS makes sure that the information you send from your computer to a website is unable to be read by any entity except you and the website you want to reach.
TLS is essential for online banking, purchases on e-commerce sites, and sensitive communications between individuals. In addition to encryption, TLS also identifies the parties involved (in this example, you and the website) so bad actors can’t get in the middle and intercept your communications.
SHAKEN and STIR offer similar protections. SHAKEN defines the framework within which STIR can function. They work together in your phone system to identify callers and make sure that calls on a SIP session originate from an approved location.
How Does This Affect VirtualPBX?
The robocall spam you receive happens from within networks that use IP-based calling mechanisms. Spammers use the same type of technology that much of the telecom industry, including VirtualPBX with its Business Phone Plans, uses to process legitimate calls.
Spammers can enter the market for an extraordinarily low cost. They can also rely on the fact that the system isn’t inherently built to stop them. There’s no mechanism in place to establish secure, verified SIP calling sessions in a way that combats spam.
SHAKEN/STIR can effectively shake up (sorry) the situation by creating a mechanism where information stored in each calls verifies the legitimacy of the caller. It will also work with non-SIP parts of calls to further validate call origination along its path.
You can read more about how the PSTN handles VoIP calls in a feature on our blog. For this robocall issue, understand that there are parts of the public phone system that use the internet to manage calls. Not all of the information used in an IP-based call is necessarily used in an analog call, but SHAKEN/STIR should have the ability to authenticate data through the entire path, regardless of the underlying platform.
Protecting the Consumer
VirtualPBX and other hosted phone service providers use carriers – including the ones the FCC addressed in its letters – to move calls between their endpoints. The adoption of SHAKEN/STIR methods across the board will be good both for individual consumers and our business customers.
When we hand off calls to carriers, we can be sure that their internal processes verify the authenticity of those calls. This situation will hurt spammers’ ability to flood our customers’ phones with robocalls, and it will create a more secure network for calling overall.
We’re excited to see what the telecom industry will do with these new spam-fighting processes. And we certainly hope that, in the coming months, the number of robocalls inundating the public will shrink and disappear.
Prepare Your VoIP Phone System for the Holidays
Posted on November 13, 2018 by Casey Houser
The holiday season is in full swing. For all the small businesses and enterprises out there: It’s time to winterize your VoIP phone system.
Today is the day to update your Automated Attendant, sweeten your Customer Greeting, correct those Holiday Hours, and change some individual Voicemail greetings to let customers know when and how you’ll be available in the coming weeks.
This post will lead you through our Dash Business Phone System settings so you can easily make your changes before the customer rush.
Automated Attendant and Customer Greeting
Your Automated Attendant, if you use one, will face your customers before anyone else they encounter in your VoIP phone system. It greets everyone with important information such as your hours of operation and a friendly hello.
In all cases, you will want your Attendant to be personable but brief. When recording its introduction, don’t just launch into “Press 1. for Sales, 2. for….”
Take time to say “happy holidays” so your callers feel welcome. And don’t forget to update your mention of business hours, since they might change for the days surrounding Thanksgiving, Christmas, and any other holidays you celebrate this time of year.
To access your Automated Attendant in Dash:
- Click Main Number in the left-hand menu on your dashboard
- Click the Incoming Call Handling sub-menu
- Select the Virtual Receptionist link in the Open Hours tab
In the new window that pops up, you can change your Text to Speech dialogue, record your own new greeting over the phone, or upload a file from your computer.
This is also the time to change your menu selection where customers can dial to reach specific departments. If you’ve added a temporary department for holiday product returns, for example, you can add a selection for that group.
And if you don’t have an Automated Attendant selection for your holiday hours, this makes a great opportunity to add one.
Holiday Hours
Mentioning your abbreviated or extended hours in your Automated Attendant is only the first step. You can use the Holiday Hours feature in Dash to route your callers based on when you’re open or closed.
In your Dash Dashboard:
- Click Main Number in the left-hand menu on your dashboard
- Click the Office Hours Strategy sub-menu
- Select the Custom Office Hours radio button, and update your times
- Click the Office Holidays sub-menu
- Update the days your office is closed
Make sure to also return to your Incoming Call Handling sub-menu where you can select the Open Hours, Lunch Hours, After Hours, and Holidays tabs and tell the system how to route your calls during those times.
During your open hours, for instance, you may want you calls to go directly to a Ring Group. But after hours, you might want to move callers directly to the Automated Attendant.
Individual Voicemail Greetings
All the users in your Dash VoIP phone system have the ability to make their own Voicemail greetings. Your Automated Attendant might only speak for the business as a whole, but your individual employees can give callers a unique message for their statuses.
Any employee can update their voicemail: 1. Dial *97 2. The system will prompt to record a greeting, or the user can press 5 at the main voicemail menu 3. Record a greeting when prompted, then press 1 to save, 2 to listen, or 3 to re-record.
During the next several weeks, employees may be out of the office more than normal. Or their personal schedules might not fit exactly with company hours. A personalized holiday voicemail can clear up any inconsistencies.
Your personal greeting can also be a warm reminder that you will return the caller’s call soon.
Is Your Holiday VoIP Phone System Ready?
Changing these Dash settings shouldn’t take more than a few minutes. Yet they can offer a host of benefits for your business and your customers.
Updates to your greetings and business hours are essential when customers want to reach you or learn how you’ll function throughout this busy season. Don’t keep them in the dark. Take a few minutes to update your system, and everyone will be happier for it.
How the PSTN Interacts With Your VoIP Calls
Posted on November 8, 2018 by Casey Houser
If you take a look at our What is VoIP? guide, you’ll read about how the Public Switched Telephone Network (PSTN) comprises a host of analog and digital systems like cellular networks, undersea fiber optic cables, and copper telephone lines that allow people across the globe to complete voice calls.
The PSTN plays a primary role in many of the calls you make during your voice over IP (VoIP) calls, too. In fact, if your calls reach a residence or business with a landline – which a U.S. Center for Disease Control study showed that about 45 percent of households had in 2016 – you will use the PSTN to make those connections.
Let’s take a look at how your outbound business calls to customers’ landlines and cell phones would use the PSTN to transmit calling information.
About Switching Centers, Quickly
Before we get started, it’s important to know that the PSTN uses switches to transmit calling information in its network of hardware.
Calls in the PSTN often first make their way to a central office. This switching center is located close to where a call originated, such as a home phone subscriber’s residence.
Then those calls are commonly sent from the central office to a gateway — also known as a tandem office. The gateway switch is typically connected to multiple phone carriers’ interconnects – such as AT&T or CenturyLink – so carriers can route calls into each other’s local networks.
You may remember the switchboards that required individuals to manually connect incoming calls to their destinations, as is shown in the image above. Today, we have replaced those jobs with digital switches, which are just electronic devices that could fit on your desk.
The digital switches mentioned previously — the central office and gateway — make it possible for switching in the PSTN to take place not necessarily in entire buildings but in rooms that contain multiple pieces of automated telephony equipment.
The word “gateway” can refer both to a centrally-located office and the piece of telephony equipment that processes calling signals. Interconnects are a type of gateway hardware.
Business VoIP Call to Home Landline
With that information about switching centers in mind, imagine this first situation as one where your business is calling a customer’s home landline. You will call outbound from your VoIP desk phone that’s connected to a hosted VoIP service like our Dash Business Phone System.
First, the phone number you dial on your VoIP phone will make its way through the internet to VirtualPBX, your hosted phone system provider. The provider will then hand that information to a carrier, which will have to determine where to further move the call. At this point, your calling information will be in a gateway.
The gateway will move your call through one or more other gateways before it reaches a location that’s close to the customer.
When the final gateway is reached, if necessary, the carrier who moved your call to that location will hand the call through its own interconnect to another carrier’s interconnect – the carrier which owns the service the customer subscribes to. (This handoff may not be appropriate if the customer subscribes to the same carrier that moved the call to the ultimate gateway office in the first place.)
That final carrier handoff will take place at the gateway switch. Following that handoff, the gateway will then move your call to a central office.
Finally, the central office will reach the customer by ringing their home phone.
This process will work in reverse when the customer answers their phone and information is sent back through the system to you, the caller.
Business VoIP Call to Mobile Phone
What happens when you want to call a customer’s smartphone from your VoIP desk phone? Now that you understand the process of reaching a home phone, the change to a mobile phone is relatively easy.
All those previous steps are similar with the exception of one. When your outbound call finally reaches the ultimate gateway, it won’t need to seek out a central office.
Instead, the gateway will send the call digitally through the cellular network to a cell tower that rings the customer’s phone.
VoIP-to-VoIP Calls
In some cases, your calls might avoid much of the PSTN.
Consider that you, a VirtualPBX subscriber, want to call a customer that uses Vonage. It’s possible that your outbound call will reach a VirtualPBX host, connect to a VirtualPBX carrier, be routed through the internet, connect to a Vonage carrier, and finally reach the Vonage user.
That call would never touch a copper wire or a cell tower. You would have sidestepped the PSTN because you used the internet, and not traditional telephony infrastructure, to move calling information from one VoIP phone to the other.
Carriers move their information in different ways, so it isn’t always accurate to say that a VoIP-to-VoIP call would have bypassed the PSTN. However, it is possible, and with gains in VoIP use at businesses and homes, it may become more likely.
What is the PSTN? The Takeaway
What you should take away from this short lesson is that the remnants of the PSTN past are still necessary for modern communications to function. If you didn’t have central offices that transmit digital data or copper wires that carry analog signals, the communications system as we know it would break down quickly.
Keep these ideas in mind the next time you dial a customer or accept an incoming call. It’s a jungle of buildings, switching, and lines, but it all functions together to make your seemingly simple phone calls possible.
Hungry for more? Keep reading to learn about packet switching, codecs, and the entire VoIP landscape.
SIP Trunking Checklist: Are You Ready to Switch?
Posted on November 6, 2018 by Casey Houser
If your enterprise phone system is more than 15 years old, there’s a good chance you use an on-site PBX. And if you do, there’s a great chance that SIP Trunking could improve the way you make calls.
An SIP Trunking setup from VirtualPBX can save you money and increase your phone system’s flexibility. You just need to know if you’re ready for the service.
Read through this short guide to determine whether or not SIP Trunking is right for you.
Do You Have an On-Site PBX?
SIP Trunking – to be extremely brief about its setup – connects your private branch exchange (PBX) to the internet. It turns your PBX phone system into a virtual PBX service. It lets you move from your local server to the cloud.
Therefore, if you want to use trunks, you will first need an on-site PBX.
The biggest giveaway that your business uses an on-premise PBX is that you have a giant server in one of your closets. This server would connect all the lines running to your desk phones and conference phones, which could number in the hundreds.
Your IT department will know where it is.
VirtualPBX SIP Trunking works with most PBXs that are SIP-enabled, including those from vendors such as Mitel, Avaya, and Samsung. Our Sales team can help you determine if your hardware will connect to our service.
Are Your Bills Expensive and Unpredictable?
It’s likely that, if you use a PBX, you also use primary rate interface (PRI) lines. A PRI connects your PBX to the public telephone network. It gives you a dial tone, but it’s limiting and expensive.
If your enterprise has more than 50 employees, you could be paying for two or more PRI lines and spending hundreds of dollars per month to accommodate that large population.
You probably pay a telecommunications company a monthly bill for your PRIs. And you might find hidden fees associated with the use of those lines. It can add up quickly.
At VirtualPBX, we lower your monthly costs into one predictable bill, no matter how many calling channels you need.
Is Your Call Capacity Unreliable?
Depending on your setup, an on-site PBX with PRI lines can keep you from doing business as you expect.
PRI lines on your PBX can only handle a set number of calls at once – a maximum of 23 per line. If you reach that maximum, no incoming or outgoing calls can be completed. You won’t be able to reach your customers, and your customers won’t be able to reach you.
VirtualPBX SIP Trunking can handle a virtually unlimited number of simultaneous calls, and its base plans start with 20 standard channels and 5 “burstable” channels, which give you a buffer during peak hours.
It’s also easy and affordable to purchase new channels, so VirtualPBX can scale with your enterprise as it grows.
Are You Ready for SIP Trunking?
If you have an on-site PBX, your service bills are sky high, and your calling is unreliable, it’s time to switch to VirtualPBX. You can keep your existing hardware but free yourself from the cost and hassle associated with an on-premise system.
Contact our Sales team today for a quick quote.
