VoIP provides a number of advantages that make it an appealing choice for enterprises, but its success depends on the quality of your internet connection. This article will cover several strategies that you can use to improve voice quality on your network.

Latency can cause poor call quality. Implementing traffic prioritization through VoIP-optimized routers and using MPLS lines to merge connections into a software-defined wide-area network can help reduce latency.


When you place a call on VoIP, your voice signals get converted into digital data packets. These data packets are then sent across the internet to the person you’re calling. When they arrive, the data packets reconvert back into your voice signal. This happens so quickly that the process shouldn’t cause any perceptible delay in the audio. However, a delay can lead to choppy conversations, echoes on both sides of the call, and even a ‘talk over’ effect where one speaker accidentally interrupts the other without realizing it.

The underlying issue that causes these problems is often a lack of network bandwidth. You can check your internet connection’s performance using online tools to measure latency, jitter, and upload and download speeds. The ideal upload speed is three megabits per second, and the download should be around 100-300 kbps. If you find that your internet speed is lower than this, consider upgrading to a faster provider.

Other elements that can affect voice quality include the type of codec used (the device that encapsulates data and sends it over the internet) and the router rules. Your VoIP network should be configured to prioritize voice packets over other data packets so that they receive higher priority and are delivered promptly.

Another factor that can affect the sound quality is the acoustics of your workplace. Make sure that you have enough space and that your phones are positioned away from any walls or other obstacles. If you have a lot of background noise, it may be worth investing in some noise-cancellation headsets.

If your calls are prone to lag, it may be time to replace your network equipment. Try switching to a different hardware vendor or a new router. Old-age headsets and microphones could also cause the issue. You should also thoroughly inspect all cables, wires, and modems, ensuring that they are not bent or damaged.

Network latency testing, or ping tests, are a great way to see how much lag your VoIP system has and identify the source of the problem. These tests can also indicate other issues affecting your business’s VoIP service, such as jitter and packet loss. By regularly running these tests, you can ensure that your business is receiving high-quality VoIP calls at all times.


Jitter is a common issue that impacts VoIP audio. It’s caused when data packets don’t arrive in the correct sequence and can degrade the quality of a voice call. The good news is that this can be resolved by ensuring that VoIP voice packets are prioritized and sent before other data packets on your network. You can also ensure this by using an Ethernet cable instead of wireless and closing non-VoIP programs during VoIP calls.

This is one of the most important factors that affect VoIP audio quality. A high jitter can cause your voice to sound robotic, choppy, or even muffled. This is because VoIP is based on real-time technology, so arriving digital signals must be received in the proper sequence to remain comprehensible. The problem is that this requires a fast and stable Internet connection, which isn’t always possible with shared networks or dial-up connections.

Another factor that can impact VoIP jitter is the compression software used by your network. There are two types of compression: lossless and lossy. Lossless compression reduces data size without sacrificing quality, while lossy compression removes information from the original file. Both can lead to a higher amount of jitter, so choosing a compression method that doesn’t put too much strain on your system is important.

The hardware in your VoIP network can also impact voice quality. Inferior hardware may not support the audio codecs needed for a high-quality VoIP phone call or have insufficient processing power to handle the required bandwidth. You should invest in high-quality, recommended hardware to prevent voice quality issues.

In addition, your broadband connection’s stability and speed can impact VoIP sound quality. A fast, stable connection can minimize jitter and latency, while a low-quality or outdated connection can increase them. If your VoIP calls are constantly choppy or sound distorted, you might need to upgrade your Internet service provider or change the settings on your router.

In addition to these issues, you should also check your network equipment's QoS (Quality of Service) configurations. This will help to ensure that VoIP traffic receives priority over other data traffic, preventing the degradation of voice quality.

Packet Loss

VoIP converts voice signals into digital data packets that are sent over the Internet. Depending on the quality of your Internet connection, you may experience latency, jitter, or packet loss that can impact the sound quality of your calls. The good news is that several VoIP monitoring tools are available to help you evaluate and diagnose the issues.

VoIP is a complex technology that depends on a network of routers and switches to function properly. If these devices are not configured and built for VoIP, the quality of your calls can suffer. You can check for VoIP problems by running a network performance test that measures your network’s upload and download speed, jitter, and latency. These tests will allow you to determine if the problem is caused by internet instability or a faulty network structure.

One of the most common causes of poor call quality is bandwidth limitations. If your employees are using their VoIP phones to stream music and movies, play video games, and download updates on a regular basis, they may be consuming more bandwidth than your business needs for VoIP. To avoid this issue, you can restrict non-VoIP traffic during VoIP calls and perform important updates when usage is low at night.

Another major cause of choppy audio is network jitter. Jitter is the difference in the time it takes for individual packets to arrive at their destination. For VoIP calls to be clear, packets must arrive at the destination in sequence and within real-time. When network jitter becomes too high, voice transmissions are jumbled and unintelligible. To lower jitter, you can install jitter buffers that will delay sending packets until enough bandwidth is available to send them.

A final reason for choppy audio is network congestion. If a lot of information is being transmitted over the same network at once, it can overwhelm and overwork your equipment. To prevent this from happening, you can implement QoS (Quality of Service) on your routers and switches to prioritize voice packets over other types of network traffic. This will ensure that your VoIP calls can be transmitted without interruptions.

Quality of Service

When your clients call, they expect to be connected with you without delay or interruption. The sound quality of VoIP calls is one of the most important aspects of the customer experience, and even minor issues can cause a lack of brand trust and communication failures. It is, therefore, crucial to regularly test your network to identify problems and find solutions before they affect your business. This includes checking jitter, latency, and packet loss.

The most common causes of poor audio are a low internet connection or insufficient bandwidth, both of which can be remedied by ensuring your routers are optimized for VoIP, investing in a high-quality headset and VoIP-optimized router, and restricting downloads during your call times. In addition, many issues affecting VoIP sound quality are caused by network congestion, and QoS (Quality of Service) can help to prevent this by prioritizing voice traffic on your network over other applications.

VoIP packets are transmitted over a LAN and WAN, which means they compete with other data traffic on your network. This can result in delayed or choppy audio, which is often due to network jitter. Jitter is a variation in the arrival time of coded speech packets, and varying routes of the individual packets, contention, or network delays can cause it. Jitter is a problem for VoIP because it disrupts the re-creation of the audio signal and leads to a jumbled, choppy, or distorted sound.

Another common issue that can disrupt VoIP call quality is dropped calls. This can be a result of network congestion, overdue software updates, or insufficient bandwidth. If you’re experiencing this issue, it’s recommended to look at your bandwidth usage and limit other online activities, increase the amount of data you can transfer per second, or use a cloud-based VoIP provider with unlimited data transfers.

A third common issue is jitter, which occurs when coded speech packets arrive at their destination with gaps in the sequence. This is due to a difference in the timing of each packet or a variation in their route over the internet. Typically, you should aim to keep your jitter below 15-20 milliseconds in order to have good VoIP call quality.