Lo-Fi Phones and the Future

bossanovalithium writes "Back in 1936 — 74 years ago — boffins accepted that about 3.3Khz was the accepted frequency that telephone calls are going to run on and it's been like that, generally, ever since. Call quality is reasonable but leaves a lot to be desired. Think calls from Skype to Skype where quality is often crystal clear." It's crazy to me that (for people with decent mics at least) Ventrillo sounds better than corporate conference calls.

Comments

guess what

[Anonymous Coward]

I live in 3rd world country and our major cellphone networks support hd-voice codecs.

Re:guess what

[sakdoctor]

Only HD?
You're all a bunch of suckers buying HD phones and HD TVs, when Ludicrous Definition is just around the corner.

Re:guess what

[Anonymous Coward]

Ludicrous Definition? Pffft.

Here in Scotland, our TV definition has gone plaid.

Coral Cache ....

[Qubit]

Just click here and avoid the Slashdotting...

Bandwidth not Frequency

[RichMan]

I can't even read the referenced article but I can tell you the phase ""Back in 1936 — 74 years ago — boffins accepted that about 3.3Khz was the accepted frequency that telephone calls are going to run on" is totally wrong.

What they meant to say was that the relevant bandwidth for understanding speech would be from 100Hz to 3.4kHz. Making the required bandwidth be 3.3Khz.

Re:Bandwidth not Frequency

[avandesande]

Damn it, 3.3Khz is more Khz than anyone will ever need!

Re:Bandwidth not Frequency

[Gordonjcp]

Actually the bandwidth is 3.1kHz, running from 300Hz to 3.4kHz. This is the range of frequencies that conveys the most information relevant to intelligibility. Anything else makes it easier to recognise the speaker but doesn't make it easier to understand them and can make it harder to understand in noisy environments.

If you low-pass filter speech below 3.4kHz then mostly you only lose the high frequencies in sibilants, but you also filter out a lot of background noise. If you're really interested you could set up a media player to play recorded speech through a tunable bandpass filter and see what you can filter out before the speech becomes hard to understand. Once you've got a feel for how the filters affect the intelligibility, try playing it back in a noisy environment (or mix in a recording of the inside of a car or something).

The 300-3400Hz filter is pretty standard in communications, and it crops up in all sorts of places. I wrote a software-defined radio app that defaults to 300-3400 but is easily tunable up to 15kHz for either low or highpass (although if you highpass filter at 15kHz you won't hear much). Occasionally I'll use it to roll off above about 2.8kHz to remove high-pitched squeaky noises from adjacent transmitters.

Obligatory screenshot: http://www.gjcp.net/~gordonjcp/lysdr.jpg

You can see the yellow strip representing the passband of the filter. The (fairly weak) signal shown doesn't really have any strong components much above about 2kHz, and I could reduce the noise by sliding the leftmost (lowpass) filter in a bit. A quick explanation of what you're looking at - that's a spectrum plot of a chunk of the 7MHz amateur band, with lower frequencies on the left as indicated by the scale at the bottom. Since on 7MHz we use lower sideband (LSB), the higher audio frequencies correspond to lower RF frequencies (further away from the red tuning cursor).

Re:Bandwidth not Frequency

[jmv]

Actually the bandwidth is 3.1kHz, running from 300Hz to 3.4kHz. This is the range of frequencies that conveys the most information relevant to intelligibility. Anything else makes it easier to recognise the speaker but doesn't make it easier to understand them and can make it harder to understand in noisy environments.

That's a myth, 3.4 kHz is not high enough to tell the "f" sound from the "s" sound over the phone. Similarly for "v" vs "z" and a bunch of others. If phones were that intelligible, people wouldn't have to say "a as in alpha, b as in bravo, ...".

Latency?

[Moridineas]

My experience with Skype, VOIP, and even to a lesser degree cell phones is that they all have latency worse than landlines. Is this actually true?

We were considering switching our business phone lines over to Time Warner voip. I talked to one of their people on the phone. My side was landline, theirs was time warner voip. The delay was awful. We kept talking over each other. If that's the best Time Warner can do, I was very not impressed, and as a result was still have our more expensive landlines.

Is there anything to my complaint, or have I just had bad luck??

Re:Latency?

[Anonymous Psychopath]

I've been working with VoIP in enterprise environments for a little over a decade. Latency is indeed a real issue and has to be considered, however it's not as restrictive as you might think. Generally speaking, if your ping is 150ms round trip you will not be able to distinguish a delay during an audio conversation, unless you're in the same room with them. Latency up to 300ms round trip is generally considered acceptable.

Cell phone conversations may or may not utilize VoIP during some legs of their calls. If they do, it's not between the phone and the tower unless you're using Skype or some other 3rd-party application on the phone. There is a distinction between encoding/decoding analog voice and how the digital signal is transmitted; you cannot consider cell phone calls to be synonymous with VoIP even though they do share some characteristics.

While cell phones do have highly variable horsepower in the CPU, the encoding/decoding is handled in purpose-built hardware chipsets, not on the CPU. It's unlikely that the type or brand of phone has any but a negligible difference in latency. Most people do not notice the latency in cell-to-cell conversations, so it may be that you're more sensitive to it for some reason.

Another factor is that some of the widely-deployed audio codecs used to compress voice were built and tuned for English speakers. Those speaking very dissimilar languages, such as Mandarin, may find that audio quality is poorer even on the same equipment.

Lastly, there are defined codec standards for wideband audio. Cisco has been including them on all their phones for several years; I assume other VoIP manufacturers have as well but do not have personal knowledge. I found that some customers did not like using them, as they are accustomed to hearing some level of white noise in the background and are prone to misinterpret a period of silence as call disconnection. If you've ever asked "are you still there?", the clarity of the call was greater than you expected or, possibly, wanted. Even with normal quality codecs we've had to inject comfort noise for years.

Little of the above applies to video. That's a whole different story.

Re:Latency?

[natehoy]

Latency of the call is highly variable, and dependent on two factors:

1. How much latency is in the network?
2. How much latency is introduced by the VoIP conversion itself?

I joined Vonage about 5 years ago. On my first ISP, I got a little over 3/4 second of latency on a really good ISP connection. This was annoying, but not enough to really make me want to spend two and a half times as much for a landline with a non-portable number. Eventually, Vonage went through a stretch of upgrades to the firmware on my adapter and the latency dropped to about 1/4 second (all but unnoticeable). However, I traveled a lot a couple of years later and found that hotel connections tend to have a lot more latency, so I got a cheap prepaid cell for when I was on the road. I settled down to a local job again and had a lot of trouble with my new ISP for a while, resulting in poor call quality and very high latency, then we got that straightened out and I was back to 1/4-second delay, which was pretty much the rule until my company issued me a cell phone with unlimited minutes, so I ditched my Vonage line because I didn't use it. But friends who have joined since have reported very low delays, almost unnoticeable, as long as their connections were good.

So the technology has improved, but you are still dependent on someone who gives you the better tech, and on a good Internet connection between you and the adapter on the other end where the call is bridged back to a POTS network.

However, landlines have a few features that people have a hard time giving up. Whether you are willing to pay for them is a different matter.

1. No need to manage power to a device. If the wires are up, the connection is the telco's responsibility.

2. Real, honest 911 with pretty much 100% accurate location awareness. Your tax dollars at work (which are a generous chunk of the difference between telco and VoIP).

3. "Feedback loop" (you can hear yourself talk in your earpiece). This helps regulate your volume, which is why people tend to talk louder into cellphones (they don't get that feedback).

4. No-delay talking. When telcos use VoIP, they use really high-end gear and fast networks to support it.

5. True DTMF support. This has gotten a little better, but VoIP for the most part can't carry DTMF tones to sufficient clarity, so your local VoIP adapter has to recognize an attempt at one and generate a fresh tone that your analog gear can recognize. Conversations with people can occasionally be interrupted by a "BEEP" as your VoIP adapter misidentifies a sound in their voice or the background as a DTMF tone and faithfully reproduces it, and you may get occasional complaints of the same issue on the other side . If it fails to reproduce when needed and you run a menu system, your customers will really hate traversing your menus.

The net result of all of that is, well, you get what you pay for. Telcos are expensive, but you are pretty much guaranteed a good call every time. Most of the gear you probably own was built to analog specifications, and the telcos are good at maintaining that spec.

For most of us, cell or VoIP is sufficient. We're OK with slight delays, a less-than-perfect reproduction of our voices, the occasional errant DTMF tone, etc.

If you run a business and you strongly feel that clear telephony is a vital part of your business, then it's probably worth paying for in your case, or at least paying for a REALLY good Internet connection and high-end VoIP gear, not consumer-grade stuff. Though you could always run one VoIP line for a while and see how it works out (just use it for less critical calls to start with).

Re:Latency?

[Mr 44]

An MIT student did his thesis on Voice vs. Data lantency on cell phones, you might be interested in his methodology and results:
Quality of Service Analysis for Audio over Cellular Voice Networks

How many died?

[0racle]

So how many boffins died to bring 3.3Khz to our phones?

worst summary ever?

[Surt]

This has got to be up there in the competition. Doesn't layout a summary of the article. Offers an opinion about some piece of software I've never heard of. No hint of whether or not there's a proposed solution.
Bizarre.

Noooooo!!!!!!

[NixieBunny]

Don't do it. The FCC changed the way television works, and look what we have now... none of my old TVs work anymore! I dread the day when my 1936 Western Electric 202 desk set stops working just because some kid wanted to listen to his girlfriend yammer in Hi-Fi.

Re:Pardon me, but...

[fremsley471]

Pretty common, over a jar in the boozer just last night, the chap opposite blurted it about Heath Robinson idea that the local Plod had developed.

Ignore the slashvertisement for XConnect

[guruevi]

the problem has been solved yet not been implemented widely. It's called ENUM and freely available and open. No need for proprietary XConnect stuff to implement this functionality, it's based off DNS and thus already has a widely available penetration. All people (and large corporations) need to do is actually use it.

Right on the spot

[Gruturo]

I was pondering this exact stuff just today at work, since a phone call sounded kinda crappy, barely acceptable until I needed to involve 2 more people and put it on speakerphone, it became so bad we had to give up. I dropped the phone call, switched to skype, and damn what a big difference. The crappiness of POTS is ridiculous indeed, and although I see the need for compatibility, it can't die soon enough.

By the way, if you like Ventrilo, try Mumble, which, apart from being free and open source, which can't hurt according to the /. crowd, has really awesome sound quality, and you can setup your own private instance in minutes. Plus, for the MMO crowds, it has extremely low latency, awesome echo echo echo echo cancellation and built-in auto volume normalization (helpful when That Loud Guy Without Headphones keeps pressing his PTT and everyone's in pain)

It's a real problem.

[Animats]

While bandwidth is low, that's not the big problem. Quality is really hard to fix over networks with time jitter. Which is why VoIP and cell phone voice quality frequently suck. The best phone audio today is from an ISDN phone to an ISDN phone - end to end uncompressed full duplex digital with hard bit timing synchronization. (ISDN voice never caught on in the US, but it's widely used in some European countries.)

Wire-line telephony is 8 bits sampled at 8KHz, so the highest potential bandwidth is 4KHz. Compare CD audio, 16 bits sampled at 44.1 KHz per channel. Cell phones are worse; they're usually compressed down to 9600 baud or so. There are some high-end video conferencing systems with higher-bandwidth audio, but they're rare.

WB-AMR is comming to your mobile

[s52d]

Hi!

In a year or two, most GSM/W-CDMA networks will be upgraded to WB-AMR codecs.
Orange is already using it in Moldova and London, others are testing.
It is marketed as High Definition Voice.

WB-AMR uses 16 kHz sampling instead of classic 8 kHz . Together with better voice compression,
higher quality of voice is using same capacity (say, 12.2 kbit/sec) as we use today.
Of course, PCM is out.
Both sides of connection must support WB-AMR, and everything in between as well,
so for few years it might not be available across different networks.
If one terminal can not use it any more (maybe due to handover to GSM cell not supporting WB-AMR),
fallback to AMR/EFR is made on both sides, using 64k/56k PCM inbetween.

Technology is avaialble for quite same time, but terminal vendors are slowing it down.
Some 20% of all terminals have to support it, otherwise it makes no sense for operator
to buy all SW needed to implement it network wide.

Funny: good old GSM will soon get higher voice quality as ISDN.

73

Bandwidth isn't today's biggest problem with calls

[MadCow42]

I don't have an issue with the frequency range, but certainly do with latency, and the lack of true duplex any more!

I find (found) that talking on a true analog line is MUCH easier than any digital line today - be that Skype, cell phones, or even land lines in most countries. I'm always amazed when traveling abroad when I make a local call on a truly-analog system how much nicer the experience is!

With today's systems in "Westernized" countries, you can't even have an effective 2-way conversation. The duplex performance sucks - you can't hear anything while you're talking. Add to that a small but noticable delay, and you have to resort to long pauses between sentences to ensure you don't talk over one another.

Am I the only one that notices this? It's AWFUL compared to what it was like 20 years ago.

MadCow.

Let's Make it Narrower!

[Bruce Perens]

Codec2 is a digital voice codec for ham radio and potentially all low-bandwidth voice communication. Currently it fits in 2550 bits per second, and we expect it to get narrower. See the Alpha Release Code.