VoIP call quality: 9 ways to improve quality issues

VoIP call quality: 9 ways to improve quality issues

Why is Voice over Internet Protocol (VoiP) call quality important? Let's say you’re in a virtual meeting over VoIP, pitching to potential customers what could be the biggest deal of your career. It’s going splendidly—you hit every mark, never stumble, even work in a joke and time the punchline perfectly. But as you’re wrapping up your pitch, someone cuts in: “Hey, you’ve been frozen for the last couple of minutes. Can you start from the top?” 

Ouch.

You’re a victim of poor VoIP quality, and while you could blame the internet connection or your internet service provider (ISP), there are actually a few factors within your control—or your company’s—that could be affecting your call experience. 

Understanding these factors and managing them effectively can help ensure crystal-clear calls so you don't miss important conversations or risk losing clients over poor VoIP call quality.

How VoIP data is transferred

To understand what might be going wrong with your VoIP calls, it’s important to first know what VoIP is and how data is transferred on a VoIP call. 

VoIP calls convert sound waves into digital signals that are broken into packets for transmission. Then, they're reassembled and reconverted into sound waves at the other end. VoIP call quality problems arise when there's a problem with packet transmission. Understanding how digital packet transmission works can help troubleshoot VoIP call quality issues.

Landline analog calls vs. VoIP digital calls: continuous signals vs. binary signals

VoIP calls travel over the internet, unlike traditional landline calls, which use the public switched telephone network (PSTN). PSTN connections typically rely on fixed wires to transmit analog signals, where an electrical signal represents a continuously varying property of the original soundwave. 

For instance, voltage can represent acoustic pressure. Analog signals can vary continuously across a range of values, but factors like heat fluctuations, current spikes, and chemical impurities can interfere during transmission, causing hissing noise and soundwave distortion that alter sound quality.

In contrast to PSTN's reliance on analog signals, VoIP uses digital signals that encode soundwave information into binary on or off values. To convert soundwaves into digital signals, VoIP connections use an analog-to-digital converter (ADC). The ADC samples an analog audio signal and transforms it into digital output. This process allows for more precise representation and processing of the original sound, resulting in high-fidelity transmissions with minimal noise and distortion. Digital signals also require less bandwidth since they can be compressed without sacrificing sound quality.

VoIP digital audio signal transmission steps: converting, sending, and reconverting data packets

VoIP transmits digital audio data through a three-step process:

1. When you’re on a call, the data is broken into small chunks called voice packets to make the data easier to transfer.

2. The packets are individually routed on different paths to make the data transfer as efficient as possible.

3. Once all of the data packets arrive at the destination, they are put back together.

All of this should happen in milliseconds (ms) to deliver your conversation to the receiver in real time. But dropped calls, frozen screens, and cutting out every other word tell us a different story: something isn’t working within the milliseconds timeframe.

What causes poor VoIP quality?

Before diving into how to test VoIP call quality and improve your VoIP calls, it’s crucial to first identify what’s causing your issues. There are a few culprits responsible for poor VoIP quality. The big ones are jitter, packet loss, latency, and network congestion. To simplify, you can think of these as packets arriving out of order (jitter), not arriving at all (packet loss), arriving late (latency), or arriving all at once (network congestion).

Jitter

Does it sound like someone is stuttering, or perhaps their speech is completely incoherent? This is usually a symptom of jitter.

Jitter occurs when data packets arrive out of order (e.g., packet 3 arrives before packet 1) or when there are inconsistent intervals of time between the arrival of the different data packets. So, for instance, data packet 3 arrives within 10 ms, but data packets 1 and 2 take 50 ms to arrive. Because all packets must be present to transfer the data to the recipient, delays in the arrival of the packets can cause high jitter.

In other words, let’s say you’re on a flight to Bangkok from New York, and you have two layovers before you arrive in Bangkok—one in London and another in Istanbul. You have to land at each airport on time to make the last flight from Istanbul to Bangkok. If one plane is delayed, a wrench is thrown into your entire itinerary.

Similarly, if too many data packets are delayed or arrive at inconsistent times, it breaks the transfer of data, resulting in packet loss and stuttered or garbled speech.

For a more technical dive into jitter, read our article, What is jitter?

Packet loss

When a packet doesn't make it to its destination and is then dropped, it's known as packet loss. Because you need all pieces of the packet to transfer the data, if one piece doesn't arrive in time, the whole packet is lost. If you’re having a conversation where the words are cutting out, this is likely due to packet loss.

A number of issues can cause packet loss, including network congestion, outdated hardware, or overloaded devices. For a full deep dive into packet loss, check out our article Understanding Packet Loss and How to Fix It.

Latency

Have you ever been on a call where someone’s response occurs a second or two late, often interrupting another person? That's a case of high latency. Latency is the time it takes for data to travel from the sender to the receiver. Jitter and latency are often related since the arrival of the packets can affect the delivery of information.

Learn more about latency, the causes of latency, and how to reduce it.

Network congestion

If we were to point a finger at any one factor that causes jitter, latency, and packet loss, network congestion would get the blame. Like a traffic jam, network congestion is when too many requests are made at once, causing the data packets to get backed up. This often occurs when multiple users move to remote work (hello, COVID) or when a company grows substantially without updating its hardware or network capabilities.

If you’re working from home and your partner or roommate is streaming movies while you’re on a VoIP call, it’s a safe bet that your network is getting overloaded and causing your poor call quality.

To help you resolve the poor VoIP call quality caused by jitter, packet loss, latency, or network congestion, try implementing a few of the tips below.

How to improve VoIP call quality

Now that you have an idea of what’s causing your poor VoIP experience, let’s dig into nine actionable steps you can take to remedy your call quality.

1. Monitor call quality

Since there are many reasons why your calls could be going awry, it’s valuable to monitor your call quality. Twilio, for instance, offers Voice Insights to all VoIP users to track jitter, packet loss, latency, and metrics like the length of the call. 

By monitoring your VoIP call logs, you can reduce time troubleshooting call quality issues and proactively address any problems before they become detrimental to your business.

How to monitor VoIP quality

To check VoIP quality, you can use a tool like Voice Insights to perform a packet analysis of your network's voice traffic and run a VoIP quality test. The Voice Insights dashboard provides insights on key performance indicators, measuring variables that affect network quality, including:

• Packet loss 

• Jitter 

• Latency 

The Voice Insights call logs view allows you to drill down into data subsets of calls with similar issues. For instance, you can study all calls involving packet loss to look for patterns. Results can also be exported in comma-separated value (CSV) format.

2. Increase bandwidth

If your network is not large enough to support multiple devices and users, it may be worthwhile to increase your bandwidth. Higher bandwidth allows your network to handle more simultaneous connections, reducing the likelihood of congestion and improving overall performance.

This can be especially beneficial in environments with a high volume of users or devices, ensuring smoother data transmission and better VoIP call quality. By accommodating more traffic, your network will be less prone to issues like lag, dropped calls, or poor audio clarity.

3. Upgrade your router

Most routers for home or for small business use are pretty simple—you plug it in and don’t think about it again. But if you’re a growing company or having challenges with VoIP calls, it may be time to upgrade. 

Upgrading your router provides more capabilities, like prioritizing VoIP traffic through Quality of Service (QoS) settings, implementing a jitter buffer, and segmenting voice traffic with a virtual local network (VLAN). If you upgrade your router, make sure it's enabled to call via Session Initiated Protocol (SIP) and VoIP. (More on SIP here.)

4. Set up a jitter buffer

A jitter buffer helps smooth out packet distribution by collecting, storing, and sending the packets at even intervals. The jitter buffer may add a delay because of the time it takes to store and process the packets, but it can lead to a smoother experience for the end user.

5. Configure QoS

Try configuring your QoS settings to prioritize VoIP traffic. If there's network congestion, VoIP traffic will be sent first, whereas web traffic, for instance, could be deprioritized. Web pages may take a bit longer to load, but at least your call won’t cut out.

6. Segment traffic with a VLAN

Another option to prioritize voice traffic is to set up a VLAN. A VLAN allows a group of devices to share a connection to specific servers even if they aren’t in the same geographical area. This can be really helpful if your company has multiple locations or operates a VoIP call center with remote workers managing customer calls.

With a VLAN, you can segment voice traffic so that it's prioritized across users. Most enterprise networks offer VLAN, so ask your network if a VLAN can be configured for your company.

7. Convert to ethernet

If you continue to have issues with an unstable internet connection, it may be time to switch from Wi-Fi to ethernet. An ethernet cable should give you a much more secure connection compared to a wireless network.

8. Purchase a high-quality headset

As simple as it sounds, a poorly designed headset can cause all sorts of shenanigans with your voice calls. From echoes to static to sound clarity, a headset can make or break your call experience.

9. Switch off Bluetooth

If you’re using a lot of Bluetooth devices, it could be affecting your connection. Wireless headphones, mouses, keyboards, etc. can all hamper your device’s internet connection, so turn off any wireless items that you’re not using.

Experience seamless VoIP calls—start free with Twilio!

We hope these tips help improve your VoIP phone calls and prevent the stuttering, echoes, or missing words that can make VoIP conversations a hair-tearing experience. Remember, when troubleshooting VoIP calls, you should monitor your call quality to help you identify the cause of your phone issues. You can then use that information to best identify the appropriate actions to remedy your VoIP call trouble.

If none of those steps help, then consider switching VoIP services. Twilio Programmable Voice allows you to make, receive, and monitor calls with our Voice API. Get started with a free account to check it out for yourself!