How HD Audio Will Transform VoIP

Those of us old enough to remember when phones were analog and switches were mechanical have a soft spot for the clarity of the old school system. There was a distinctness and intimacy to those old time conversations that has long been lost. Well, perhaps lost only for some decades. There’s a new move afoot to give VoIP phone conversations a starting fidelity that will make today’s handsets and cellphones seem like toy walkie talkies by comparison.

The technology is HD or high definition audio. It’s called HD Voice by Polycom, the telephone manufacturer producing IP telephones with this capability built-in. The idea is to reverse the steady decline in voice intelligibility that we’ve experienced with the digital transformation and create a new standard that is superior to anything that has come before.

There’s a parallel here. Something similar happened in the radio industry. If you’ve ever owned one of those big console radios from the 1930’s or even the table models from the 50’s, you know that AM radio can sound amazingly lifelike. The advent of transistor radios shrunk the size of the loudspeakers, and thus the range of sound frequencies that can be heard. “Tinny” was the word to describe the loss of low end or bass notes. Car radios added their own restrictions by rolling off the high frequencies to reduce noise and static. It took a new technology, FM radio, to expand the frequency range and eliminate the background noise before the term high fidelity could be applied.

Old-timey phones had very few parts. The carbon microphone and electromagnetic earphone were powered by a battery at the phone company and connected to a similar phone at the other end. There were no electronics in the circuit. The telephone company switch was really a mechanical switch that simply connected phones together. The resulting sound had a richness that made local conversations much closer to someone sitting next to you than you hear today.

I remember when I first noticed the difference. It was when the company I worked for upgraded their old phone system with a new digital PBX and desksets. The first calls seemed slightly more muffled, less sparkling clear on the new phones. The effect was subtle, but noticeable. There was no noise or distortion apparent, but something was missing.

What was missing, I later figured out, was some of the audio range in the conversation. It wasn’t digital that degraded phone audio specifically. It started with long distance calling. The analog carrier multiplexing used to load multiple conversations onto copper wire and microwave trunk lines needed filters to keep one conversation from interfering with another. After much research, a frequency range of 300 to 3400 Hz was selected as standard. Frequencies above 3400 Hz add to naturalness, but don’t limit understandability of speech very much. Get rid of them and you can transmit more telephone channels on a trunk. You also get rid of annoying hiss, which is a high frequency phenomena. Frequencies below 300 Hz also aren’t necessary for intelligibility, but certainly give voices and other sounds their distinctiveness. By getting rid of the low frequencies, problems with power line hum getting into phone lines are greatly reduced or eliminated.

This standard was maintained when the industry changed from analog to digital. It kept everything compatible and consistent. Digital technology was a boon to long distance calls, especially overseas. It totally eliminated the noise and crosstalk that were common in the analog era. That advantage went away with cell phones, which can be noisy or even drop out.

Early implementations of VoIP only made matters worse. Packet transmission added distortion from lost packets and variable transmission time or jitter. Latency, a time delay through the network, was almost unknown with analog and TDM telephony. Now it’s not uncommon on poorly implemented systems to have a clipping effect so that you can’t both talk at the same time.

What HD audio intends to do is to lift telephone voice quality to a new standard above and beyond what we’ve ever had. This is accomplished by using a wideband CODEC (Coder/Decoder) to handle the conversion from analog microphone and earpiece to digital transmission. Bandwidth is doubled to 7 kHz with the new standard G.722 Codec and can be extended up to 20 Khz, the same quality as CDs and digital FM radio.

HD Voice is here now. You can get it with Polycom business phones and a service provider like RealLinx that can provide the hosted PBX and carefully managed bandwidth to maintain high quality phone audio. Of course, this only works within an organization that is set up for HD Voice or among users that all have the G.722 Codec available. When one party is using traditional telephone technology, everything defaults to the universal G.711 telco standard and its limited frequency range.

Even so, higher fidelity within your organization can improve conversation intelligibility, including nuances in the voice that are often lost when talking on the phone rather than in person. This can be valuable when budgets and time restrictions reduce face to face meetings in favor of audio and video teleconferencing.

Are you ready to replace an aging phone system or order equipment for a new enterprise or new location? If so, why limit yourself to yesterday’s technology when you can have something that will provide a competitive edge at lower overall cost than you have now? Don’t make a move until you get pricing and features on HD Voice Hosted PBX VoIP telephone systems.

VoIP That Sounds Better Than TDM Phone Calls

There is a conventional wisdom that says VoIP phone calls will always sound worse than traditional circuit switched calls using the time-honored TDM transmission system. What's more, the impression is that technology limitations ensure that VoIP can never even equal the voice quality of TDM calls. What if I told you that VoIP can sound at least twice as good as TDM and perhaps even good enough to transmit high fidelity music? Would you buy that? Actually you already can.

Astounded? I'll bet. The image of IP telephony has been severely tarnished over the years. It started with PC softphones using dial-up Internet services to make computer to computer phone calls. As long as this stayed in the realm of a hobby or a means of conversing around the world for free, it was OK. The fact that that the voices sounded like they were coming from the moon was less important than avoiding outrageous international phone rates. Everything was fine until low-end Internet Telephony started being sold as a replacement for "real" telephone service.

Even over broadband connections, poorly implemented consumer and business VoIP solutions can leave a lot to be desired. The all too familiar clipped conversations, vocal distortion, echoes and even dropped calls are seen as a necessary evil that comes with lower cost phone services. What gets lost is the fact that VoIP telephony offers opportunities to improve phone system performance, not degrade it.

All of this has made some businesses goosey about using VoIP. Even new IP PBX installations implement VoIP only for calls within the organization. A TDM interface card terminates the calls to the public switched telephone network before they leave the building. Sip Trunking from enterprise level providers such as BandTel is now convincing corporate telecom managers that they can maintain TDM voice quality while saving money over terminating calls themselves. But what about actually improving voice quality? Is this for real?

It is, but it's not widely deployed as of yet. The secret is in the CODEC you are using. A CODEC, coder / decoder, is the chip that takes an analog voice signal from your handset and converts it to and from digital packets that can be conveyed over a computer network. The grandfather of all CODECs is G.711, an ITU standard that was developed to implement digital trunking over 50 years ago. This is the one that all phone companies use to transmit long distance calls. It's also the default CODEC for digital handsets, including IP phones. Why? Because you can be sure that no matter who you call, they have G.711 available.

G.711 is called the "toll quality" standard CODEC. That's because it replaced existing analog carrier telephony standards with the same bandwidth per channel. That bandwidth is 3 KHz, ranging from 300 to 3300 Hz. You wouldn't be impressed with an AM radio that had a frequency response of 300 to 3,300 Hz, but this is what we've come to accept as the gold standard for telephone calls. Sometimes this narrow bandpass is sold as being tuned to enhance voice signals, since it does tend to filter out low level power line hum and high frequency hiss. But it also cuts out so much of the vocal range that it can be hard to understand some speakers.

What's better? A wideband CODEC gaining acceptance is G.722. It's also an international standard. Wideband refers to the fact that G.722 offers a frequency response of 50 to 7,000 Hz. That's better response than many small radios and enough to make voices sound natural, not "tinny."

Oh, but how much bandwidth is that going to consume on the network? The same, actually. The bandwidth for G.711 is 64 Kbps. The bandwidth for G.722 is also 64 Kbps. How can that be? Compression. G.711 is an uncompressed datastream. G.722 requires more sophisticated processing to cram more vocal range into the same bandwidth. Think of it a little like MP3 encoding for music. With clever algorithms, you can eliminate redundant and unnecessary data bits that won't be noticed by the listener.

How can you improve the voice quality of your VoIP phone system? If your handsets and system support G.722, use this CODEC instead of G.711. If your system only supports G.711 or the lower bitrate G.729 CODEC, you'll need to upgrade to experience wideband voice. Polycom is a leader in improving VoIP sound quality, calling their approach HD Voice. They implement a standard variation of G.722 plus careful voice processing for echo cancellation and noise reduction in their handsets and systems. You can listen to recorded samples of voice and music that make the improvement very apparent.

Implementing HD Voice or G.722 within your company gains the immediate benefit of less fatiguing, easier to understand phone conversations. As long as your telephony remains packetized, such as over a point to point T1 data line to another branch office, you maintain the advantage of wideband voice signals. The fly in the ointment is that inevitable connection to the outside world. Right now all calls will be converted to the G.711 standard as soon as they touch the PSTN. And there goes the quality improvement. Only by using a voice and data extranet to your customers or suppliers, or by sharing the same SIP trunking provider that keeps your calls as packetized voice, can you currently enjoy extended quality telephone calls outside your company.