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HD voice is offered by most IP PBX and Unified Communications (UC) vendors as well as some Session Initiation Protocol (SIP) trunk providers. You may already own HD voice. HD voice is a capability existing in many IP phones and softphones but enterprises may not know they can turn it on. Implementing HD voice has value, even though there can be some compatibility issues, and it's limited to IP networks.
What is HD Voice?
Enterprises deploying IP PBX and UC systems could be overlooking the HD voice feature. TDM phone systems limit the audio frequencies carried to a range of roughly 300 to 3400 Hz (narrowband). HD voice offers wideband capability, carrying analog frequencies from 50 to 7000 Hz. This analog bandwidth is converted into a digital stream of 64 kbps or less. There are even audio codecs that can deliver CD quality sound 50 to 22000 Hz digitized at 64kbps, 96kbps, and 128kbps.
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HD voice calls are conversations where both ends have HD/ wideband equipment. The connection between them supports up to 64 kbps digitized speech. HD conference calls are where at least some of the participants are connected in wideband through a wideband-capable bridge. The conference call sounds much better than over a public switched telephone network (PSTN) connection. When you use Skype, you are experiencing HD voice. HD voice has been announced on Sprint's wireless network.
The Value of HD Voice
So what are the benefits of HD voice?
- Recognize the speakers more easily
- Understand speakers who use English as a second language with accents that can be difficult to understand
- Overcomes background noise during the call
- Reduces audible mistakes (Mark Stratton of Siemens reported at an Enterprise Connect conference that HD voice could reduce the listening errors from 40 down to 4 per 30 minute conversation)
- Less strain/stress occurs with HD voice when there are long conversations
- Improves the sound quality for conference calls
In addition to the "soft" benefits of wideband noted above, there can be direct cost savings as well. Connecting calls via IP avoids PSTN and conferencing charges.
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Not All of HD Voice is the Same
Enabling wideband/HD calling with an IP-PBX is usually a matter of setting a configuration parameter as long as both ends of the call have HD voice capable IP/softphones. The cheapest IP phones usually do not have support for HD voice. HD voice does not require any more bandwidth than a standard G.711 digitized call operating at 64 kbps.
Most IP PBX and UC systems can support a common wideband codec, ITU standard G.722. This codec (coder/decoder) operates with 64 kbps or less digital bandwidth. There are numerous other HD codecs including G.722.1, G.722.2, Speex, iSAC, SILK, iPCM-wb, EVRC-WB, most which are not supported by SIP trunk providers or Session Border Controllers (SBC). These codecs are incompatible with each other.
Microsoft supports its proprietary softphone codec, RTAudio with the Lync Server. RTAudio dynamically adjusts itself based on existing network conditions. Check with your SBC vendor and SIP trunk provider to determine if they support RTAudio. When RTAudio leaves the IP network, it is transcoded (converted) into G.711 and does not carry HD voice.
HD Voice Limitations
The challenge is that HD voice calls cannot be routed through the public telephone network, PSTN. The trunk gateways to the PSTN require the use of the G.711 codec standard. HD voice stops at the gateway boundary of the IP network.
Once the call exits the IP PBX HD voice island, then supporting HD voice becomes an issue. A few SIP trunking providers can support HD voice when SIP trunks are implemented at both ends of the call. A few Session Border Controllers (SBC) can support HD voice like AudioCodes and Cisco. However, most SBCs do not support HD voice. When a SBC receives HD voice and the SIP trunk does not support HD voice then the SBC must transcode the HD voice into G.711 or G.729 for transmission over the SIP trunk. In the near future, SBCs will be able transcode RTAudio (from Microsoft) into G.722 for SIP trunks.
Making HD Voice Work
There is concern that VoIP calls will be of lower quality as packets get lost or delayed. Experience has shown that call quality can be delivered as long as an enterprise has a "right-sized" broadband connection. Right-sized means that there is sufficient bandwidth available for the maximum number of anticipated simultaneous phone calls in addition to data traffic, and the router supports Quality of Service (QoS) for voice on IP networks. When packets enter the backbone network, current infrastructure typically delivers zero packet loss even on the Internet.
The IP PBX routing instructions should connect calls via IP, not the PSTN, either using the public internet or peering trunks between IP PBX's or SIP trunks. Connecting to a wideband conference service is easier. When an IP PBX extension dials the service's 8XX number, the IP PBX should route the call directly to the conference bridge in IP, rather than sending it via the PSTN.
You Should Try HD Voice
If you have not tried out your own HD voice within your IP island, do so. Listen to the difference. Once the user has HD voice exposure, it will be hard to take it away. HD voice does not require any more digital bandwidth than a narrowband call. HD voice does not tax the IP network any more than a standard digital voice call. The concept of HD voice conferencing extends the HD quality beyond the enterprise for connection to other HD islands. If you are operating in Europe, you will find HD voice more frequently implemented.
Source: http://www.webtorials.com/discussions/2012/10/getting-ready-for-hd-voice.html
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