Getting Ready for HD Voice

High Definition (HD) voice sounds great (pun intended).  In general, HD voice might more appropriately be called "HD telephony" since most of the technical discussion here is about the audio portion of a call, which is generally audio-only.  However, the same technology is equally applicable for multimedia conferencing.
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.
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.
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.

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In the section on HV voice transcoding, you mention G.711. G.729, and RTAudio. To what extent do these codecs “cover the waterfront”? Are there additional codecs that should be included for wider interoperability?

Most SBCs and SIP trunk providers cover the codec s for G.711, G.729 and RTAudio. Additional codecs that be offered may include G.723, G.728, GSM, iLBC, AAC-LC and AAC-LD.


As a daily user of HD voice (via Lync and the Microsoft RTAudio codec), I can testify to virtually all that was said above. Once you go HD, reverting back to narrow-bandwidth G.711 or other low bit-rate codecs is almost painful.

The next frontier is HD audio to mobile handsets. This is much of what is driving the voice of LTE push, but with wide-band wireless codecs. (what more codecs?!?!)

Interestingly enough, interoperability between the wireline and wireless application of HD telephony has been held up by the need for low-cost and low-latency transcoding.


Great article (and comments). I wanted to offer some feedback on the SBC capabilities touched on in this article. I do think the SBC vendors (in particular my employer Acme Packet) offer many more capabilities in the area of HD voice and associated transcoding requirements then everyone may realize. As an example, Acme Packet has been supporting HD voice since 2010 and has one of the broadest set of HD voice & transcoding capabilities in the industry. This includes support for the vast majority of codecs referenced in this article. The demand for SBCs to support HD voice and transcoding will likely remain strong because just as death and taxes are always certain, there is also certainty that the industry will never converge on a common codec.


As we noted in our June 2011 byline article on the role of SBCs in enabling HD voice (, an SBC must be able to perform real-time transcoding between different HD voice codecs during a VoIP call, must support media pass through capability (this eliminates or reduces the signaling loss due to transcoding), must support dynamic reordering of codecs during a VoIP call setup, must support multi-party call that involves a mix of narrowband and HD participants, and must provide intelligent routing of HD voice calls through the network (i.e, route HD calls through HD network route, if available, to avoid downgrading to narrrowband)which is usually accomplished with a policy engine that supports intelligent routing.
Sonus is at the forefront of enabling HD voice in VoIP/SIP networks. Sonus offers a portfolio of SBC products that supports most of these capabilities. In particular, the Sonus solution includes a robust and field proven policy engine that performs intelligent routing of HD sessions, and the SBC HW architecture is designed to scale without compromising session capacity - this is an important consideration for real time HD transcoding.
HD voice/video technology is here to stay. With the deployment of LTE/VoLTE, the need for "HD enabled" SBCs and other network elements is likely to increase in the coming years.

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