Another aspect that reinforces your point is that the ATM push (and subsequent downfall) was not just bandwidth-motivated but also motivated by a belief that ATM's QoS guarantees were necessary. But it turned out that software improvements, notably MPLS to handle QoS, were all that was needed.
Plus the cell phone industry paved the way for VOIP by getting everyone used to really, really crappy voice quality. Generations of Bell Labs and Bellcore engineers would rather have resigned than be subjected to what's considered acceptable voice quality nowadays...
3G networks in many European countries were shut off in 2022-2024. The few remaining ones will go too over the next couple of years.
VoLTE is 5G, common throughout Europe. However the handset manufacturer may need to qualify each handset model with local carriers before they will connect using VoLTE. As I understand the situation, Google for instance has only qualified Pixel phones for 5G in 19 of 170-odd countries. So 5G features like VoLTE may not be available in all countries. This is very handset/country/carrier-dependent.
Yes, I think most video on the Internet is HLS and similar approaches which are about as far from the ATM circuit-switching approach as it gets. For those unfamiliar HLS is pretty much breaking the video into chunks to download over plain HTTP.
Yes, but that's entirely orthogonal to the "coding" algorithms being used and which are specifically responsible for the improvement that GP was describing.
HLS is really just a way to empower the client with the ownership of the playback logic. Let the client handle forward buffering, retries, stream selection, etc.
>> Plus the cell phone industry paved the way for VOIP by getting everyone used to really, really crappy voice quality
What accounts for this difference? Is there something inherently worse about the nature of cell phone infrastructure over land-line use?
I'm totally naive on such subjects.
I'm just old enough to remember landlines being widespread, but nearly all of my phone calls have been via cell since the mid 00s, so I can't judge quality differences given the time that's passed.
Because at some point, someone decided that 8 kbps makes for an acceptable audio stream per subscriber. And at first, the novelty of being able to call anyone anywhere, even with this awful quality, was novel enough that people would accept it. And most people did until the carriers decided they could allocate a little more with VoLTE, if it works on your phone in your area.
> Because at some point, someone decided that 8 kbps makes for an acceptable audio stream per subscriber.
Has it not been like this for a very long time? I was under the impression that "voice frequency" being defined as up to 4 kHz was a very old standard - after all, (long-distance) phone calls have always been multiplexed through coaxial or microwave links. And it follows that 8kbps is all you need to losslessly digitally sample that.
I assumed it was jitter and such that lead to lower quality of VoIP/cellular, but that's a total guess. Along with maybe compression algorithms that try to squeeze the stream even tighter than 8kbps? But I wouldn't have figured it was the 8kHz sample rate at fault, right?
Sure, if you stop after "nobody's vocal coords make noises above 4khz in normal conversation", but the rumbling of the vocal coords isn't the entire audio data which is present in-person. Clicks of the tongue and smacking of the lips make much higher frequencies, and higher sample rates capture the timbre/shape of the soundwave instead of rounding it down to a smooth sine wave. Discord defaults to 64kbps, but you can push it up to 96kbps or 128kbps with nitro membership, and it's not hard to hear an improvement with the higher bitrates. And if you've ever used bluetooth audio, you know the difference in quality between the bidirectional call profile, and the unidirectional music profile, and wished to have the bandwidth of the music profile with the low latency of the call profile.
> Sure, if you stop after "nobody's vocal coords make noises above 4khz in normal conversation"
Huh? What? That's not even remotely true.
If you read your comment out loud, the very first sound you'd make would have almost all of its energy concentrated between 4 and 10 kHz.
Human vocal cords constantly hit up to around 10 kHz, though auditory distinctiveness is more concentrated below 4 kHz. It is unevenly distributed though, with sounds like <s> and <sh> being (infamously) severely degraded by a 4 kHz cut-off.
AMR (adaptive multi-rate audio codec) can get down to 4.75 kbit/s when there's low bandwidth available, which is typically what people complain about as being terrible quality.
The speech codecs are complex and fascinating, very different from just doing a frequency filter and compressing.
The base is linear predictive coding, which encodes the voice based on a simple model of the human mouth and throat. Huge compression but it sounds terrible. Then you take the error between the original signal and the LPC encoded signal, this waveform is compressed heavily but more conventionally and transmitted along with the LPC signal.
Phones also layer on voice activity detection, when you aren't talking the system just transmits noise parameters and the other end hears some tailored white noise. As phone calls typically have one person speaking at a time and there are frequent pauses in speech this is a huge win. But it also makes mistakes, especially in noisy environments (like call centers, voice calls are the business, why are they so bad?). When this happens the system becomes unintelligible because it isn't even trying to encode the voice.
The 8KHz samples were encoded with relatively low encoding complexity PCM (G.711) at 8KHz. That gets to a 64kbps data channel rate. This was the standard for "toll quality" audio. Not 8kbps.
The 8kbps rates on cellular are the more complicated (relative to G.711) AMR-NB encoding. AMR supports voice rates from about 5-12kbps with a typical 8kbps rate. There's a lot more pre and post processing of the input signal and more involved encoding. There's a bit more voice information dropped by the encoder.
Part of the quality problem even today with VoLTE is different carriers support different profiles and calls between carriers will often drop down to the lowest common codec which is usually AMR-NB. There's higher bitrate and better codecs available in the standard but they're implemented differently by different carriers for shitty cellular carrier reasons.
> The 8KHz samples were encoded with relatively low encoding complexity PCM (G.711) at 8KHz. That gets to a 64kbps data channel rate. This was the standard for "toll quality" audio. Not 8kbps.
I'm a moron, thanks. I think I got the sample rate mixed up with the bitrate. Appreciate you clearing that up - and the other info!
And memory. In the heyday of ATM (late 90s) a few megabytes was quite expensive for a set-top box, so you couldn't buffer many seconds of compressed video.
Also, the phone companies had a pathological aversion to understanding Moore's law, because it suggested they'd have to charge half as much for bandwidth every 18 months. Long distance rates had gone down more like 50%/decade, and even that was too fast.
