Before we’ll sell you a single Cloud PBX line, Intermedia will ask you to take a free voice quality test. If you’re talking to other providers, you’ll note that our test is more comprehensive and longer-lasting than the rest.
That’s deliberate. In order to help assure the best experience for people calling on your network, we’re going to put it through a stringent testing process. This is how we help ensure excellent VoIP calling quality even when your network is congested from file downloads or video streaming.
What is voice quality testing?
Our voice quality test helps make sure that your network can support the additional load caused by VoIP phone traffic.
In the testing process, we’ll drive rigorous voice and data traffic to your network to simulate real-life loads.
To perform the test, we will do the following:
Set up a phone simulation software on your network
Tune the software to the number of phones you are looking to buy
Run the simulation as if all phones are being used for three full days, in parallel with your regular network tasks
Capture all network traffic information and analyze it
More rigorous testing than most other providers
We hold ourselves to an extremely high standard. One of the key differences in how we test is that we’ll perform our simulation over three full days. This is important because the traffic on your network often varies depending on the time of day and even the day of the week.
For example, think about when you perform system backups or transfer big files to your customers. These tasks create congestion in your network that could impact voice calling quality. And a short test will not detect them.
What we’re testing for
Our engineers analyze the stream of data packets (those bits of digitized information that flow through the network) along the following criteria:
Bandwidth. We’ll verify that your actual bandwidth is sufficient for both data and voice traffic.
Packet loss. If packets are dropped on their way to their destination because of network congestion, there will be interruptions to your conversation. We need to spot it ahead of time.
Latency. This is when packets take too long to get to the other side, which causes delays in the conversation. Any delay over 60 milliseconds is noticeable and disruptive.
Jitter. Jitter is when data packets arrive in different order or at a different pace from when they were sent. If not corrected, jitter causes a noticeable drop in the quality of the phone call.
Complete connectivity disruption. A complete disruption of the network connection will naturally cause calling problems.
How much bandwidth do you need for Cloud PBX?
Generally, a Cloud PBX VoIP call will use a G729 codec or approximately 30 kilobits per second (kbps) of bandwidth for the duration of the call. If you were to add Cloud PBX for 5 users, and everyone was on the phone at the same time, it would require 150 kbps of bandwidth during the calls. (For reference: 1,000 kilobits is equal to 1 megabit.) Your exact needs will depend on the call volume and call patterns at your company as well the type of IP phones you are using. You can use your existing internet connection for Cloud PBX as long as the connection is dedicated (not dial-up) and you feel you have adequate upstream and downstream bandwidth for the number of users you are adding.
What if we find problems?
In more than 90% of the tests we perform, we find no issues.
But when we do find problems, we do two things: first, we recommend that you hold off on your buying decision until you adjust your network environment. And second, we formulate a plan to fix the issues.
In some cases, this is as simple as adjusting your backup schedule. In other cases, we need to work with your internet provider—and we’ll actually join calls and reach out to the provider on your behalf to sort out the technical issues.
For partners: test before you sell with VoIP Scout
Intermedia’s VoIP Scout enables Cloud PBX and SIP Trunking resellers to conduct comprehensive self-service testing of their customers’ data networks in order to identify and mitigate potential issues that could impact voice quality or system performance.
VoIP Scout lets partners simulate the fullest potential load the VoIP services could conceivably put on the network over 3 to 5 business days. The scope and duration of the test allows partners to inspire confidence in customers that may otherwise be wary of VoIP or proactively address network issues that could prevent high-quality VoIP service. Intermedia’s VoIP Scout consists of three core components:
The VoIP Scout Appliance for running tests via an inexpensive piece of dedicated hardware (ideal for locations with no available PCs and/or dedicated voice networks)
The VoIP Scout Soft Client for running tests via on-site computer in the background
The VoIP Scout Management Portal for scheduling, reviewing and managing network tests
The VoIP Scout Soft Client and Management Portal are available free for all partners. The VoIP Scout Appliance is sold to partners at cost.