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VoIP audio quality help

How to get the best out of your VoIP service.

Audio quality during a call

The best way to make and take a VoIP call is on a physical VoIP phone – using its handset. The handset has noise cancelling which efficiently reduces background noise. This is done by having a microphone pointing towards your mouth and another microphone pointing away from your mouth. When a sound enters both microphones, one microphone has its audio inverted and added to the the audio coming in through the other microphone effectively cancelling out the sound common to both microphones. Therefore theoretically, highlighting the sound coming from the main microphone.

The problem is that if you point the main microphone away from your mouth or if you hang the handset on your shoulder, your voice may become garbled.

If you need to use a speaker phone, please be aware that the microphone at your end may pick up unwanted sounds and the phones noise cancelling may cause your voice to burble.


Wired analogue (which connect to your PC via a 3.5mm jack) have been shown to be the best type of connection.

USB connected headsets come next for quality but they are very close to analogue headsets.

The worst type of head set are the wireless ones. The radio signal causes loss of quality in both directions and the fact that the microphone is at the earpiece and not near the mouth at all will cause loss of quality.

Mono or Stereo?

This is up to personal choice. Full stereo systems use double bandwidth but most are just mono split into 2 speakers.

Quality of audio

There are several techniques used to convert audio to a digital format which can be sent up and down the internet. The techniques are optimized for quality, audio bandwidth, compression rate, error checking etc.

Older digitizing techniques used analogue sampling and sent the resulting digital signal down the line. The larger the sample rate the better the quality.

The techniques are referred to as codecs and good ones usually cost extra due to licencing.

The audio bandwidth is very important for audio quality and it is possible to have to wide a bandwidth. The maximum sensible bandwidth is 64k and is used by the OPUS codec and very few others. Seen to be excessive and roughly equivalent to 4K in terms of a TV.

Some handsets such as the Yealink, will have this built in. If you phone one OPUS enabled device from another OPUS enabled device the codec may flick automatically up to OPUS – ie the highest quality supported by both devices.

The most popular codecs use 32k and classified as HD quality.

The higher the audio bandwidth, the more data needs to be sent and the more processing power is needed to compress it and decompress it at the other end.

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