The switch to a VoIP phone system can be tricky. A strong, reliable internet connection isn’t the be-all, end-all to a seamless experience. If your installation isn’t properly configured and if your hardware isn’t up to snuff, you’ll find that VoIP issues keep cropping up and hindering your ability to place calls.
We know that this is frustrating and confusing. We don’t want it to be. So we rounded up the usual suspects of VoIP issues and created a lineup to help frustrated and confused VoIP users everywhere.
Problem: VoIP issues and poor call quality
To understand what’s going wrong, first let’s look at the ideal VoIP setup. In a nutshell, VoIP calls happen like so:
One user speaks into a computer or headset microphone. That person’s voice is digitized into an audio stream, and split into packets of data. Those packets travel through the user agent’s computer (or softphone, or smartphone, etc.), router, and Aircall’s providers’ media server. Then the packets travel through several VoIP and traditional telephony carriers, then through the end user’s installation. The end user listens, then speaks, and the process is repeated in the other direction.
This long journey is rife with opportunities for things to go awry. Users can’t really step in and fix issues pertaining to the carrier or the other person on the line. The orange box on the left displays the potential hiccups on an agent’s network. This article will help you understand and diagnose those local VoIP issues and get the most out of your installation.
So, what’s going on?
VoIP issues can render calls anything from annoying to impossible. The symptoms can take many forms: echo, popping, distortion, one-way audio, dropped calls, etc. When your call quality suffers, it may be due to one or several of the following VoIP issues:
Audio latency can happen during two operations: during the time it takes to encode the audio, and during the time the packet takes to travel through the agent’s network. Latency doesn’t affect the audio quality of the call, but rather its intelligibility. As latency increases, the call will quickly become impossible to understand as the two participants will start to talk over each other.
Jitter happens when too marked a discrepancy occurs in the rhythm of packet delivery. Audio must be played at a constant rhythm to be intelligible, so if packets aren’t delivered at a constant rhythm, jitter will be noticeable. VoIP services include jitter buffers, but sometimes latency will go beyond the buffer’s capacities. If this happens, the end user will notice missing or ‘skipped’ audio (if the packets are being delivered too quickly) or lapses of silence (to account for slow packet delivery).
- Packet loss
If packets are lost, delayed, or contain errors, they might be dropped and abandoned before reaching the end user. This is most often due to an overloaded network or unreliable connection. The result is missing chunks of audio.
These disturbances are perceived by the user as distorted, missing, or choppy audio. This guide aims to help you find the source of the disturbance, and remedy it through better network and hardware configuration.
A perfect storm
These issues usually happen in conjunction, and result in poor overall call quality. Given the demanding nature of VoIP, your network could be suffering from these issues, and you wouldn’t know until you tried to place a call. A high-speed fiber-optic internet connection isn’t everything, unfortunately. Your network needs to be properly configured in order to get the absolute most out of your VoIP software.
Sorting out and optimizing your VoIP network is a serious endeavor. It requires know-how and planning. But that fact remains, until you take the plunge, your call quality issues may never fully disappear. Luckily, this investment of time and resources is a one-time deal: once you’ve made it, you’ll see results right away, for the better.
Configuring your network
Like we mentioned above, high-speed internet isn’t enough to guarantee flawless call quality. Of course, without sufficient bandwidth, VoIP calls won’t go through. But even with lightning-fast internet speeds, you could still be encountering VoIP issues. This guide is here to help when you do have the right setup, but VoIP issues still plague your calls.
Voice packets travel through many stages before they find their way to the person on the other end of the call. This includes access points, routers, switches, and more. Every point on the voice packets’ journey is a potential bottleneck and source of VoIP issues.
Working with your IT team to smooth out the interaction of switches and routers and how they process data is a necessity. There are several parameters to take into account: your total volume of calls, the number of concurrent calls, activity spikes, etc. You’ll likely need to tweak your network installation, but again, once you iron out the kinks, you will notice a definite upswing in call quality. This task is daunting, but crucially important to your call quality.
Network prioritization is your friend
Network prioritization sounds intimidating, but it’s actually your ace in the hole to do away with VoIP issues. When users on your network consume bandwidth, all their applications contribute to traffic on your network, like cars driving next to each other on a wide, busy street.
However, without proper prioritization, all the cars will get in each other’s way. Some vehicles like firetrucks won’t be able to speed down empty lanes in order to fulfil their time-sensitive purpose. VoIP calls are firetrucks of sorts, in the sense that they’re very sensitive to network traffic. For this reason, they need to be prioritized before other bandwidth-hungry applications, such as your email client or music streaming service.
This prioritization is called Quality of Service, or QoS. It will ensure that voice packets have a higher traffic priority than other data packets. Your IT team can take care of configuring your router, and once it’s set up, it’s set up for good. It’ll protect your call quality by making sure that your calls always have the appropriate bandwidth set aside for their needs.
Check out the latency and packet loss improvements one of our clients achieved after setting up QoS on their router:
Their situation wasn’t catastrophic, but issues routinely got in the way of their full enjoyment of their phone system. Here’s what happened after QoS:
Don’t use wi-fi (unless it’s airtight, which it likely isn’t)
If your users’ computers are using wi-fi, they’re more likely to experience call degradation and VoIP issues. The reason for this is that your router is probably not configured for 100% coverage all over your offices. This is likely for three reasons:
- First, a router of this kind is expensive and needs to be intentionally configured.
- Second, wi-fi was never designed with real-time applications in mind.
- Third, there is bound to be interference on your wifi network, since almost every office abounds with electronic devices which can disrupt one another.
Using wi-fi for VoIP calls can work perfectly fine, but it’s probably not worth the headache. An ethernet hookup is a much safer choice. It will ensure a constant, higher quality network connection.
Use the right hardware
The previous section dealt with configuration and internet connection. The other usual suspect for VoIP issues is the voice equipment on the user’s end (at the very left of the diagram above).
Interference happens when another electronic device hinders a VoIP call. This could result in popping, crackling, or a hum. Wireless devices such as smartphones, microwaves, and fluorescent lights are often at fault here, as is using a wi-fi connection instead of an ethernet cable.
Using wired devices instead of wireless ones is always a good move, and could drastically reduce VoIP issues. Try troubleshooting if you’re finding that one hardware setup refuses to work properly even though everything else is on the up-and-up. Relocate the call and switch out your hardware to find the culprit.
You might not pin them as troublemakers, but headphones are responsible for more VoIP issues than you suspect. The reason for this is that, as headphones become more technologically sophisticated, they also become more complex. Manufacturers make headphones drastically different from one another, often with irritating consequences for the end user.
Every provider of an app or program that uses sound must contend with capricious headphone specs, and VoIP users are somewhat in the same situation. Some headphones will disconnect if the user stays silent for two long. Others will refuse to be recognized by certain operating systems. Others still will cause feedback as a user’s voice goes out a speaker on the other end of the call, and is picked up by the end microphone and sent back to the original user. This can cause sound distortion or one-way audio.
This kind of vexation is why many providers will be happy to recommend a professional VoIP headset which they know won’t cause trouble for their users (our go-to is the Jabra Evolve 40).
This guide isn’t exhaustive, and one of the properties of VoIP issues is that they can be tough to replicate, diagnose, and fix. We want each and every one of our users to have an easy time using Aircall, which is why we’re always trying to improve. If you’re encountering an issue we didn’t detail, feel free to check out our FAQ or reach out to our support team, firstname.lastname@example.org.