How to troubleshoot VoIP call quality issues
I think we’ve all been in a position lately where a VoIP call sounds pretty bad. But what if it’s happening all the time?
While VoIP typically offers similar or even better call quality to PSTN or legacy PBX calls, quality issues can still be a pain for your average user.
So here’s a quick rundown from your favorite cloud comms experts, starting with an understanding of what you’re experiencing.
Typical VoIP call quality issues
While it can make more sense to just say, ‘Jessica’s voice keeps skipping on our call’ or ‘the guys from accounting sometimes sound like robots’, it’s important to know the jargon so that you can better identify the issue.
Things you want to look out for when it comes to your voice quality are the likes of:
- Latency – The lower the latency, the more your call will replicate the experience of two people talking as if they are in the same room. As latency increases, you’re likely to be left with those uncomfortable interruptions and pauses that make people talk over each other, then awkwardly apologize.
- Jitter – In effect, jitter is the change in latency over time. Jitter will be especially apparent if your call needs to traverse the public internet. It usually sounds like interference or like someone is trying, and failing, to plug in their mic.
- Packet loss – When voice signals are digitized and transmitted, they are split into packets. Some of these packets fail to reach the endpoint – small pieces of the audio signal will be missing, resulting in audible distortion on the call. This is your classic case of robot voice.
So now you hopefully know the technical name for your problem, let’s get down to how you’re actually going to go about fixing it.
Step 1: Check your bandwidth
It might seem obvious to some but if you’re going to be moving your voice data as well as your usual internet data over the same connection, you’re going to want to make sure you have the capacity. That means checking your bandwidth. If your bandwidth is pretty limited you might want to consider getting a better connection.
Some applications use a lot of bandwidth, especially file streaming apps. Try closing applications like Netflix, Spotify, YouTube, etc. It also might be a good idea to run a test for malware and/or spyware on your PC/s that could be slowing your connection.
Step 2: Make sure QoS is enabled
This is a bit of manual work you might want to take up before trying out SIP trunks. Don’t worry, it’s not too technical, and you shouldn’t need to move from your desk to get it done. Basically, your router has this setting called Quality of Service (QoS). Turning this on means that your router will prioritize your voice traffic rather than your regular internet data. What this will do is reduce latency on your calls and should help prevent your calls from cutting out unexpectedly.
If you happen to have a pretty old setup, you’re going to want to change your router out for one that has this setting or configure it yourself (good luck). That way you’ll make sure your calls go as smoothly as possible.
Step 3: Check your equipment
If you’re using old equipment it’s possible that you need an upgrade to solve your issues with things like jitter and packet loss. To do a quick check though, try switching out equipment one at a time till you see a difference. A common fix is testing out another headset or changing a wireless mic for a wired one.
You can also get an echo if your equipment is too close together, causing interference. I know it can be satisfying to have everything in one spot but if your PC, phone, router and
Echos can also be the result of electromagnetic interference created when your VoIP hardware is too close to other electrical devices. Make sure your phone, computer, power strip, router, splitter, and IAD aren’t sat on top of each other or you’ll be at risk of getting feedback.
Step 4: Check your internet speed
If your ISP is having downtime, you’ll probably find that’s what’s causing your call quality problems. Try doing a speed test and take a look at your ping, if it’s over 50ms, you’re looking at your problem. Anything over that and you’re going to get echoes, delays, and all sorts of quality problems.
If you’re working with a VoIP provider that offers a full PSTN replacement from the cloud, your voice operations will be run entirely over a dedicated data network. If that’s the case your IT team needs to put traffic management policies to prioritize real-time media including voice and video so that network congestion doesn’t harm the quality of your phone calls and conferences.
Still got issues?
If you’re still having problems and you’ve done all the above you might need a VoIP engineer to take a look at your setup so they can identify and fix your problem.