How to ensure your network can support good quality voice calls

In this digital landscape, Voice over Internet Protocol (VoIP) has become the go-to solution for seamless business communication. From audio calls to virtual meetings, VoIP offers convenience and cost savings. However, to ensure seamless conversations with crystal-clear audio, one crucial factor takes center stage: internet quality. Poor network conditions can chop your voice into a distorted mess or introduce annoying delays, hindering effective communication. In this article, we will talk about internet quality testing for VoIP calls, and equip you with the understanding to optimize your network and enjoy uninterrupted, high-quality conversations.

Testing Your Network Quality for VoIP

Audio problems experienced during connected calls are typically attributed to the quality or capacity of the network. To ensure smooth and uninterrupted conversations, it is advisable to test your network for various factors such as jitter, packet loss, latency, and bandwidth. It is recommended to conduct multiple tests as network parameters tend to fluctuate frequently. To test the quality of your network, you can use the Speedtest app, which can be downloaded for iOS, android or desktop from https://www.speedtest.net/apps.

Ideal Parameters for Clear-Audio Calls

To ensure clear-audio calls, aim for the following parameters. Just remember to conduct these network tests multiple times. Network parameters can fluctuate frequently, and by repeating the tests, you can improve the chances of catching and resolving any potential audio problems effectively.

Bandwidth: You should always maintain a minimum bandwidth of 100kbps clear up/down per call to accommodate the audio data without congestion or quality degradation. Bandwidth problems are caused by overloaded connections or insufficient network capacity.

Packet Loss: There should be no or 0% packet loss to eliminate any audio disruptions caused by missing data packets. Packet loss attributed to the intermittent loss of audio in connected calls

Jitter: Keep jitter below 15ms, with lower values preferred. This will help you maintain a consistent and smooth audio flow during conversations and avoid choppy or garbled audio.

Latency: The latency should be under 150ms to minimize delays and ensure a natural conversational pace.

If you want to clearly understand the effect of each of these parameters on the quality of a connected call, you should continue reading further.

Understanding the Parameters for Good Audio

Imagine your internet connection as the conduit through which your voice travels in the digital realm. Like any pathway, it has its limitations and potential roadblocks that can impede the flow of data packets carrying your audio. These roadblocks manifest in the form of bandwidth constraints, packet loss, jitter, and latency. Understanding and testing these parameters is key to ensuring the optimal performance of your VoIP calls.

Bandwidth

Bandwidth refers to the capacity of your internet connection to transmit data. In the context of VoIP communications, sufficient bandwidth is crucial for clear audio. Imagine your internet connection as a pipeline, and the data packets of your audio call as water flowing through it. If the pipeline is narrow or congested, the water flow will be restricted, causing audio issues. Similarly, if your bandwidth is limited or overloaded with other data-intensive activities, it can lead to distorted or dropped audio during VoIP calls. For example, downloading large files or streaming high-definition videos while making a VoIP call can consume significant bandwidth, affecting the audio quality.

Packet Loss

Packet loss occurs when data packets are lost or fail to reach their destination. In VoIP calls, packet losses result in gaps or missing segments of audio, leading to a fragmented conversation. To understand packet loss, imagine you are talking on a VoIP call, but certain parts of your sentences are not reaching the other person. This can cause confusion and make the conversation difficult to follow. Even a small percentage of packet loss can be noticeable during conversations. For instance, if 5% of your audio packets are lost, the recipient may not hear every 20th word you speak.

Jitter

Jitter refers to the variation in the delay of data packet delivery. In VoIP calls, jitter usually disrupt the smooth flow of audio, causing irregularities or distortion. To illustrate jitter, imagine sending a series of audio packets, and each packet arriving at different intervals. This inconsistency can result in audio that sounds choppy or garbled. Excessive jitter can make conversations challenging and frustrating. If you experience high jitter, your words may be heard in a fragmented or inconsistent manner, making it difficult for the other person to understand you.

Latency

Latency, often referred as Ping, is the time it takes for data to travel from the source to the destination. In VoIP communications, high latency can cause noticeable delays between the time you speak and when the other person hears you. This delay can disrupt the natural conversational flow and lead to awkward pauses. Imagine having a phone conversation where there is a significant delay between when you speak and when the other person hears your voice. This delay can hinder real-time communication and make the conversation feel disjointed. Summarizing it all, it is essential for you to test your internet quality for VoIP to ensure clear-audio calls. By understanding and optimizing the parameters of bandwidth, packet loss, jitter, and latency, you can significantly enhance your VoIP communication experience. You can use network testing apps like the Speedtest app to assess your network’s performance and make any necessary adjustments. With a reliable internet connection, you will be able to enjoy seamless, high-quality VoIP calls and avoid poor audio to facilitate effective business communication.