G.722, HD Voice and HD Audio have become the latest buzzwords in the VoIP (Voice Over Internet Protocol) market in the last year. They are all words to describe the same thing - wideband audio that delivers voice calls using VoIP with audio quality that is greatly superior that of a regular landline or mobile phone call.
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.
Sample 1: Asterisk recording "Your extension number is two two two". This file uses the GSM codec for "Your extension number is" and uses the G.722 codec for the "two two two"
Sample 2: Asterisk recording "At the sound of the tone the time will be exactly twenty fourty nine". This file uses the GSM codec for "At the sound of the tone the time will be exactly" and uses the G.722 codec for the "twenty fourty nine" CLICK HERE FOR SAMPLE 2
Sample 3: Asterisk Voicemail recording using G.729. This is a sample of the Asterisk voicemail prompts using recordings in the native G.729 format.
Sample 4: Asterisk Voicemail recording using G.711. This is a sample of the Asterisk voicemail prompts using recordings in the native G.711 format.
Sample 5: Asterisk Voicemail recording using G.722. This is a sample of the Asterisk voicemail prompts using recordings in the native G.722 format.
Sample 6: Asterisk voice call using G.729. This is a sample of a phone call using the G.729 codec.
Sample 7: Asterisk voice call using G.711 a-law. This is a sample of a phone call using the G.711 a-law codec.
Sample 8: Asterisk voice call using G.722. This is a sample of a phone call using the G.722 codec.
The differences between the various codecs should be readily apparent!
Other related posts:
Obihai OBi 200 Analogue Telephone Adapter (ATA) Review
Raspberry Pi – the ultimate home Asterisk PBX.
Linksys SPA New Zealand Configuration
Comment by gtxboyracer, on 5-May-2010 11:12
wow the G.722 codecs are amazing quality. Do we need to make sure our ATA's etc support such a codec before we can take advantage of that?
I currently have a LINKSYS (CISCO) SPA2102
Comment by AndrewTD, on 5-May-2010 12:03
Polycom is huge on HD Voice. It's one of the most exciting differentiators of their products in the market place IMO.
Here's a few links for those who are interested:
Firstly - Polycom web page on HD Voice: http://www.polycom.asia/company/about_us/technology/ultimatehd/hdvoice.html
Interesting video of Polycom Co-founder & CTO Jeff Rodman being interviewed on HD Voice: http://www.tmcnet.com/tmc/videos/default.aspx?vid=1934
It is a little "dumbed-down", but still worth watching.
And here is a white paper on the various wideband codecs. It gives an excellent technical overview of the various wideband codecs.: http://www.polycom.asia/global/documents/whitepapers/codecs_white_paper.pdf
Comment by maverick, on 5-May-2010 12:21
I will add that the WorldxChange network supports G722 across our entire network and have done for quite some time , so any VFX / DVX or direct connected user that has a device that supports G722 will be able to have HD Audio calls to any of our IP direct connected customers that also have G722 enabled devices, only when we have to go back to a leagacy TDM network will it not be able to be used.
Comment by AndrewTD, on 5-May-2010 13:51
It is the audio bandwidth aspect of the codec that makes it "wideband", not the sampling rate.
E.G. G.722 has an audio bandwidth of 7kHz; Siren 14 is 14kHz, etc.
There are three metrics that sometimes get confused when discussing audio codecs:
1. audio bandwidth - means the audio range of sounds that can be reproduced by the codec, from low pitch sounds (think sub woofer range in your home theatre) to high pitch sounds (think tweeter) this is measured in kHz. It's actually a range, but often the one figure is given to indicate the range from 0 to X kHz
2. data rate: this is often described in kbps and describes the bandwidth of IP traffic generated by the codec. Think of this in terms of how much of your IP capacity on your network will be consumed by the voice traffic.
3. sampling rate: this is how many times per second the codec "samples" the analog sound input (to then convert that sample to a digital output). This too is measured in kHz (and is thus sometimes confused with audio bandwidth)
Comment by antoniosk, on 5-May-2010 18:37
Good article Mr Biddle. TCL supplies Polycom HD phones for it's IP Gateway product set and the vox quality is very nice.
One day when the sip interconnects are built and all carriers transmit with open protocols... then the vox quality will be nice.
Comment by webwat, on 7-May-2010 15:55
How does it compare to iLBC? My phone crashes when I enable it, but supposed to be better than G.711
Comment by Brian Wade, on 19-Sep-2013 06:53
Comcast's Hosted Voice Service offers G.722!
Comment by Allan, on 28-Jul-2014 15:40
I understand that to have an mobile HD voice call both phones need to be HD capable and on the Vodafone network, however what codecs are used when an HD capable phone on the Vodafone network rings a PSTN number, such as an 0800 number or local number? Does the mobile part of the call drop down to AMR/AMR-NB or does the codec stay at AMR-WB and transcode to G711 at the PSTN inter connect?