Skype lets you do audio conferencing with wideband codecs, and a service called Vapps High Definition Conferencing does the same thing for non-Skype VoIP calls.

Now other VoIP providers can offer wideband conferencing too. A company called Wyde Voice sells an all-IP conferencing platform that natively uses wideband codecs. The Wyde platform uses the iSAC codec from GIPS, so anybody calling in from a soft phone like the Gismo5 client, or the Google, AOL or Yahoo VoIP clients can enjoy a conference in wideband. If one of the participants in the call is using a narrow-band codec, the Wyde device up-samples the signal to wideband quality for mixing.

I have always been an enthusiastic proponent of wideband audio – it is one of the major potential advantages of VoIP over circuit switched telephony. Circuit switched calls are encoded with G.711, which yields 12 bits of effective dynamic range and a maximum frequency of about 3.5KHz. Human speech has harmonics even above 10KHz, which is why it is hard to tell the difference between and “F” and an “S” over the phone. The G.711 codec places an absolute limit on the sound quality of a regular phone call. A VoIP phone call can use a wideband codec, with whatever dynamic range and frequency range you want. There are several of them, commonly with a sample size of 16 bits and a sampling rate of 16KHz which captures a maximum audio frequency of 8KHz. When you have a good enough connection Skype uses a wideband codec by default, which is why it can sound better than “toll quality” (if you aren’t limited by your loudspeaker and microphone.)

Unfortunately, for the non-Skype world there’s a chicken and egg problem – almost no phones support wideband codecs, so the carriers aren’t motivated to support them either. Worse, any VoIP call that traverses the PSTN at any point is converted to G.711, losing the wideband frequencies. Worse yet, to cut costs most carrier implementations of VoIP use a bandwidth-saving codec that intrinsically delivers inferior sound quality to G.711; for example, last I heard Vonage was using G.729A.

As VoIP matures, and more and more calls are IP end-to-end through VoIP peering and ENUM arrangements (what Gizmo5 calls “back-door dialing,”) wideband codecs will become more pervasive and our conversations will become clearer. The Wyde announcement is a step towards that world.

CSR announced their Bluecore 6 chip today. It will ship in production volumes in January 2008. CSR claims a more robust connection - with increased transmit power and receive sensitivity. CSR also claims a breakthrough in sound quality, achieved by going from a Continuous Variable Slope Delta (CVSD) codec to Adaptive Differential Pulse Code Modulation (ADPCM). This enables packet retransmission and a halving of transmission bandwidth. The reduced bandwidth requirement results in a reduction in power consumption, and the ADPCM codec yields a MOS of 4.14 compared with a maximum of 2.41 for CVSD.

This is a welcome change, but doesn’t really go far enough. What’s needed is a wideband codec like AMR-WB to yield better-than-toll quality sound. While this would be redundant in a regular cell phone - ADPCM is more than adequate to carry a signal that has been encoded in GSM - it would make a huge difference in dual-mode phones carrying Voice over Wi-Fi.

***Update: I went to the T-Mobile store this morning and signed up. The service here in Dallas is $10 per month, not $20 as reported by Reuters. The store manager also told me that people with poor cellular reception at home can use the UMA service at no additional monthly charge, but that this usage is treated the same way as cellular usage - in other words, it counts against your cellular minutes.***

***Update 2: Here are some details on the T-Mobile launch campaign. ***

Reuters reported this morning that T-Mobile is rolling out FMC service nationally.

Subscribers would pay an extra fee of up to $19.99 per line or $29.99 for five lines on top of regular monthly cellular bills for unlimited calls in a subscriber’s home or the nearly 8,500 places T-Mobile runs Wi-Fi, like Starbucks coffee shops.

This pricing model seems ambitious, compared to what it is competing with. T-Mobile’s MyFaves 300 plan gives you unlimited minutes nights and weekends and unlimited minutes to a list of five people that you choose. So the 300 minutes are consumed during the day, calling to people whom you call infrequently. For $20 more you can bump this to 1,000 minutes. Alternatively, you can spend that $20 on the FMC service. It seems like the FMC service would only be a better deal for people who are home all day (or at Starbucks), who want to talk a lot to people beyond their five most frequently called. MyFaves 1000 would be a better deal for people who want to talk to a large variety of people during the day when they are not at home, for example in the car or out of range of a Starbucks - like at work, for example.

So who are these people that this “HotSpot@Home” service is aimed at? Surely there can’t be many. Why doesn’t T-Mobile use this technology to gain more customers, by giving it away free to subscribers? This would appeal to all the people who have poor reception at home, who would feel bilked by having to pay extra just for acceptable quality of service there (Hey! They do! See the update above). Another way to increase customer appeal would be to go with a wideband codec for Wi-Fi calls, guaranteeing CD-quality sound to Wi-Fi on-network calls. Or why not do both? This would provide a viral motivation to complement MyFaves, it would be unique among US carriers, it would improve retention, and it would bring new subscribers to start exploiting all that spectrum that T-Mobile picked up in the AWS auction in September 2006.

Potentially VoIP calls can sound radically better than what we are used to even on landline phones. So why don’t they? It may be lack of will. Some say the success of the mobile phone industry proves that people don’t care about sound quality on their calls. I don’t think this is a valid inference. All it proves is that people value mobility higher than sound quality.

The telephonic journey from mouth to ear, often thousands of miles in tens of milliseconds, traverses a chain of many weak links, each compounding the impairment of the sound. First, the phone. Whether it’s a headset, a desk phone or a PC, the microphone and speakers have to be capable of transmitting the full frequency spectrum of the human voice without loss, distortion or echo. Second the digital encoding of the call; it has to be done with a wideband codec. Third, the codec has to be end-to-end, so no hops through the circuit switched phone network. Finally the network must convey the media packets swiftly and reliably, since delayed packets are effectively lost, and lost packets reduce sound quality.

Discussions of VoIP QoS normally dwell mainly on the last of these factors, but the others are at least as important. The exciting thing about dual-mode cell phones is that they provide a means to cut through them. Because they must handle polyphonic ring tones and iPod-type capabilities, the speakers on most cell phones can easily carry the full frequency range of the human voice. Cell phone microphones can also pick up the required range, and DSP techniques can mitigate the physical acoustic design challenges of the cell phone form factor. Smart phone processors have the oomph to run modern wideband codecs. This leaves the issue of staying on the IP network from end-to-end. The great thing about dual-mode phones is that they can connect directly to the Internet in the two places where most people spend most of their time: at work and at home.

So if you and the person you are talking to are both in a Wi-Fi enabled location, and you both have a dual mode cell phone, your calls should not only be free, but the sound should be way better than toll quality.

Check out the V2oIP website for an industry initiative on this topic.