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Voice Quality in VoIP Calls

Updated on June 20, 2013

Defining VoIP Quality

A "good" quality telephone conversation is defined differently by various people. Which is why in order to get a quick overview of the quality of VoIP calls, you are simply asked to rate it on a four or five star scale. Nevertheless, it is possible to break down the concept of quality into various factors. We can talk about latency, breaks in the flow, loss of audio, and the actual reproduction of the voice itself. Different aspects of the network are responsible for determining the overall quality of a VoIP call.

The first thing to ensure is that you have sufficient bandwidth for VoIP. In today's broadband connected world, this will almost certainly not be a problem – at least in a home environment. When you're talking about a business however there may be many employees talking at the same time and the bandwidth requirements increase dramatically. Your IT administrator needs to create a separate VLAN to ensure that there is always enough bandwidth for voice calls to go through even when many people are using the system simultaneously.

Latency is another big issue. In regular conversation, the maximum tolerance for latency is around 300 ms round-trip time. But since you don't have much control over most of the parts, you need to reduce it at your end as much as possible. Having VoIP data centers close to your location and minimizing the number of steps that it has to jump through will help.

VoIP Call Quality
VoIP Call Quality | Source

Selecting a Quality Audio Codec

A hosted VoIP phones service will be able to provide you with a wide range of audio codecs to choose from. The regular PSTN codec samples your voice at 8 kHz which cuts off quite a large number of higher and lower frequencies. This makes it impossible to accurately reconstruct your voice which leads to problems such as being unable to distinguish between various fricative consonants. With VoIP, you can select a much higher quality audio codec that samples your voice at 16 kHz instead. It provides a much more accurate reproduction that is a pleasure to use and is difficult to let go once you've gotten used to it.

The true magic of a VoIP call comes when all of the components are working together perfectly. If even one of them is off, it can spoil the overall call quality. Sufficient bandwidth, good quality codecs, and low latency together combine to produce a sound quality that is far superior to anything that the PSTN phone system can compare to. Contact your SIP service provider for help in properly setting up your network. After all, they not only have a lot of experience with this, it's in their best interests to ensure that your infrastructure is working smoothly. Unlike with the regular telecom industry, there is a lot of competition in the VoIP marketplace and you can pick and choose your service provider as per your requirements.


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    • Zubair Ahmed profile image

      Zubair Ahmed 4 years ago

      Nice hub and real good info, thank you for sharing. Yes you are correct VOIP is improving and how good the voice sounds at each end depends on many factors which you have nicely provided.