Tuesday, September 18, 2007

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.

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