Since the late 1970's G.711 has been the defacto standard in the telephony world for voice encoding as we moved into the digital world with fully digital phone switches, and moved away from analogue phone exchanges. G.711 sampled audio at 8kHz and created a 64kbps audio stream using two slightly different methods depending on where in the world you were located. u-law was used in Japan and North America, and a-law was used in the remainder of the world. u-law and a-law were also used as the audio codec for both T1 and E1 circuits as well as ISDN, hence the reason that channels are all 64kbps. Since the mid 90's as VoIP has rapidly taken over in the telephony world G.711 has still remained as the codec of choice.
Things are now slowly changing however, since computers now have far more processing power than they did in the 1970's the ability to process, compress, and decompress audio in real time is now far easier. This has lead to the creation of new audio codecs in recent years that can sample voice with greater frequency ranges, and also compress audio to use less bandwidth without any noticeable loss of quality.
G.729 delivers call quality that is only marginally less than that of G.711 but uses approximately half the bandwidth. This offers very significant benefits as we move to fully IP based networks as it allows greater volumes of voice traffic to be carried. G.722 on the other hand uses a similar amount of bandwidth as G.711, but samples audio at 16kHz which is double that of G.711 and delivers what many regard as far more natural sounding audio. Newer codecs such as Siren22 that was created by Polycom take things a step further and sample audio at 22 kHz, resulting in audio that sounds even better but with the downside of using significantly more bandwidth. G722-2 or AMR-WB is also slowly making it's presence felt in the mobile market with a number of mobile carriers having recently deployed this codec which offers superior 16 kHz voice quality over a mobile connection when the end users both have handsets that support this codec.
Many modern IP PBX's support the G.722 audio codec and with most new IP phones supporting G.722 it's certainly a feature that many people are rapidly discovering. The enhanced audio quality also makes audio conferencing a vastly superior experience. While G.722 is currently supported by a growing number of VoIP providers around the world, the one "limitation" of G.722 is that there is no benefit when calling a landline, mobile phone, or another VoIP user who doesn't have G.722 hardware as the call will only be as good as the audio quality from the remote party. Here in New Zealand WorldxChange fully support G.722 on their VFX and DVX platforms for calling between other DVX and VFX users, and G.722 calls between users on different VoIP providers who peer IP calls and support G.722 are also possible. Telecom New Zealand's new VoIP platform and newly released SIP trunking product does not support G.722 at this stage and I see this being a fundamental downside for them as they move towards launching this product publically in the coming months.
I'm presently reviewing a number of mid range and high end VoIP handsets for Geekzone. I have a Snom 820, Cisco SPA509G, Aastra 6755i, Yealink T28P and Polycom IP450, all of which support the G.722 codec. As part of this review I've recorded some audio samples and have them available below for you to listen and compare for yourself.