How to Achieve Ridiculously Good VoIP Call Quality

In today's digital world, there is a lot of information about VoIP communications available right at your fingertips; it's only a Google search away. And if you've done any research so far, you are familiar with one of the biggest concerns that business owners face when they move their phone systems to the cloud: call quality.

In this post, we'll share the secret to achieving ridiculously good call quality with your VoIP phone system. Some of the key factors we will discuss include allocating the right amount of bandwidth, using the right codec for transmitting voice traffic and adjusting your Quality of Service (QoS) settings.

Bandwidth and VoIP Call Quality

All cloud communications applications use Voice over Internet Protocol (VoIP) to send and receive calls. Because of this, your call quality is directly dependent on the Internet service that your business uses. If you do not allocate enough bandwidth for your cloud-based phone system, you will experience poor call quality.

Before you make the switch to a cloud communications solution, it is important to evaluate your current Internet Service Provider (ISP) and ensure that your network can handle the additional voice traffic.

The bandwidth needed for superior VoIP call quality depends on how many concurrent calls your business experiences during peak hours as well as the codec that your VoIP provider uses. (We'll talk more about codecs in the next segment.) For most standard codecs, you will need to allocate anywhere from 85 - 100 kpbs of bandwidth per concurrent call.

Codecs and VoIP Call Quality

Short for "encoder-decoder," a codec converts an audio signal, which is the speaker's voice, into digital form so that it can be transmitted via the Internet. Then, it converts it back into the audio signal so that it can be heard by the listener. The codec that your VoIP communications provider uses plays a big role in call quality.

Some common codecs include:


This codec has no compression. It is often the default codec for SIP trunk providers and can be used anywhere that enough bandwidth is available. This codec is often used for domestic phone calls.


Although this codec offers better call quality than the public switched telephone network (PSTN), it is not often used with SIP trunking solutions. It requires more bandwidth, but it does allow for superior call quality.


This codec compresses voice signals from 64K to 8K. It provides reliable call quality and is often used in situations when bandwidth is at a premium, such as international calling.

Be sure to ask your provider which codec they use for both local and international calls, as the compression may differ for each. For the best call quality, you will want to choose a VoIP provider that uses G.722 or G.729 codecs.

Quality of Service (QoS) and VoIP Call Quality

In most cases, adequate bandwidth and the proper codec is enough to ensure high call quality. But if you are still having problems with delay, jitter, or packet loss, you will want to adjust the Quality of Service (QoS) settings on your router. QoS settings allow businesses to prioritize a specific kind of data, such as voice calls, within their network - ensuring superior call quality. Proper Q0S settings allow your router to quickly identify voice traffic and place it in a reserved queue that takes priority over all forms of data on the network.

Learn More: Achieving Superior Call Quality

Looking for more information on how to achieve superior call quality on your VoIP phone system? Contact an AVOXI VoIP specialist today, and explore the articles below.