What’s the difference between standard and High Definion VoIP?

MURRAY LEACH

Telux HD Co-Founder and Managing Director.

When you make or receive a call over VoIP, your phone system has to transform your voice from analogue sound waves into a digital format so that it can be transported through over the phone network and changed back into sounds at the other end. This transformation is carried out using a compression algorithm known as a codec. The particular codec used determines the perceived quality of a VoIP call. So what’s the difference between standard and high definition VoIP? First, let’s dive into codecs…

What’s a Codec?

A codec, which stands for coder-decoder, converts an audio signal (voice) into compressed digital form for transmission (VoIP) or storage and then back into an uncompressed audio signal for playback. It’s the essence of VoIP. Codecs vary in sound quality, bandwidth required, and computational requirements. Each service, program, phone, gateway, etc., typically supports several different codecs, and when talking to each other, negotiate which codec they will use.

Fun Fact – You can assign a different codec to individual phones. Although the reasoning behind wanting to do so, bandwidth usage, is largely overstated. The average saving is 40Kbps per second, which in our opinion, isn’t worth the quality trade off.

Common VoIP Codec Protocols

G.729 is a codec that has low bandwidth requirements but provides acceptable audio quality. This is the most commonly used codec in VoIP calling and has an MOS (Mean Opinion Score for voice quality) rating of 3.9.

G.711 is a codec that was introduced by ITU in 1972 for use in digital telephony. With only a 1:2 compression and a 64K bitrate for each direction (128K plus some overhead), it is best used where there is a lot of bandwidth available. G.711 has a MOS rating of 4.2

G.722 can be used at lower bit rate than G.711 (48/56/64Kbps) ITU standard codec which, because it is of even better quality of the traditional public switched telephone network (PSTN), it can be used for a variety of higher quality speech applications. This standard also requires an adequate amount of bandwidth, often available in business applications, and usually rates a 4.5 on the MOS scale.

In the late 1970’s it was agreed that the standard codec for telecommunications was going to be G.711. The vast majority of phone systems today still use G.711, as it’s not only reliable but the sound quality is rather good for such an old method. G.711 has also maintained popularity as it not only has a low bandwidth requirement, but also isn’t very demanding when it comes to processing power requirements of phone hardware or routing equipment. Needless to say [almost entirely thanks to Moore’s Law], things have improved since the late 70’s. As times have changed, technology has come on leaps and bounds. Bandwidth capabilities have increased and processing power is a lot more advanced, meaning we no longer have to rely on the standard G.711.

The Telux VoIP platform uses G.722. Although G.722 is much more demanding on processors than G.711, when compared to other ‘higher spec’ codecs, it uses less bandwidth and provides a superior quality of sound. This codec not only excels when used for standard phone calls, it really shines when used for conferencing and loudspeaker calls too. It’s easy for us to write about the ways G.722 is far superior to G.711, but the best way to compare these two codecs is to listen to them for yourself.

The differences between G.711 and G.722 may seem subtle on paper, but the real proof is in the pudding. Here’s two audio clips, the first is our IVR recording using G.711 (standard definition) and the second using G.722 (HD VoIP).

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