1. I think the lines have jitter now they're over IP anyhow.
2. Bit niche...
1. I think the lines have jitter now they're over IP anyhow.
2. Bit niche...
On the other hand, it could assume much less noise than an analogue phone line.
VoIP codecs are heavily optimized for human voice and have just a few kilobits of bandwidth. The analog phone network, before it went entirely digital, had a much higher bandwidth which is why modems were able to reach 56kbps. Phone calls in the 1980s were often crystal clear, almost on par with a broadcast radio station. Cell phones today along with VoIP are a muddy unintelligible mess in comparison.
That is actually completely false. The way that speed was achieved was by using multiple bits per symbol encoding (bitrate != baudrate), not by having some mysterious unused bandwidth available. This was the case for speeds above 4800bps up to 33kbps (V.34).
56kbps (V.90) was only possible with PCM (digital) modulation, while the actual symbol rate on the line was still only about 8000Bd - i.e. about 8kHz, which is about the maximum for a typical landline.
This is all due to how a typical phone line works and that there is a very strong attenuation of higher frequencies due to parasitic capacitance and inductance of the wiring.
I am not sure where do you live and what kind landline phones you had but modern digital cell phone codecs are light years ahead of the sound quality we had on land lines in 80s, esp. when calling long distance.
There is certainly a lot more nuance to reality than my blanket statement was meant to imply but I assumed that went without saying. But I believe the underlying gist is accurate. Fancy encoding techniques got us quite far on analog phone lines, but my point was that attempting to transmit digital information over a highly-compressed audio stream optimized within an inch of its life for a small subset of human vocal sounds is going to be MUCH slower than over the analog phone lines of yesteryear. And at the end of the day, there is no substitute for bandwidth. Take it up with Claude Shannon if you like.
> I am not sure where do you live and what kind landline phones you had but modern digital cell phone codecs are light years ahead of the sound quality we had on land lines in 80s, esp. when calling long distance.
I lived in the Midwest US (and then later the West, with frequent calls across the country) and I quite distinctly remember the audio quality of POTS being quite good overall. Of course, not EVERY call was great. Some exchanges were noisy, some people's local loop or wiring in their house had issues, and some phones themselves were just garbage. But it mostly sounded great.
I'll grant that cell phone audio is more tolerable now than it WAS but it certainly was so bad for a while that from the early 2000's to the early 2010's, I pretty much only used cell phones for emergencies and actively avoided calling people on their cell phones if I could. The resulting audio was so distorted and muddy that I often could not understand the person on the other end. If it was someone I knew, I could often infer what they were saying from their speech patterns. But if it was someone I didn't know, or if they had a thick accent, there was always a lot of, "Can you repeat that again, more slowly this time?". I only cancelled my analog phone line after POTS network providers started switching their internal systems to VoIP (or something like it), bringing along all the same disadvantages of the bandwidth-constrained cell network.
Latency was also terrible, and continues to be. The analog POTS circuit-switched network was (modulo an audio buffer or two) speed of light. Modern digital voice communications buffer like crazy and up to two seconds of latency is extremely common. Maybe I'm an outlier as most people don't seem to notice but for some reason, even a half-second of latency is enough to make a conversation extremely awkward for me.
ime both can be shitty or "crystal clear" but i totally agree that having a cell phone call today, with worse quality than an cross-continent analog connection in the 80s is very depressing
If quality means low latency and low jitter, and you can get T1/E1 digital telephone service, circuit switched end to end, that's almost certainly the lowest practical latency and jitter. It'll cost real money and you might not like the 8000Hz 8-bit sampling. ISDN calling is pretty much the same thing, but you only get two voice lines instead of 24 or 25.
If that's not an option, but low latency is still important, a solid wired connection with g.711 SIP is pretty close, but packets are 20ms of samples, so that adds to your baseline latency. G.711 is basically the same codec as used for t1. And internet jitter is probably not zero, but if your connection is well provisioned it's often not that bad.
If you want a wider audio band, maybe try to get SIP with G.722.2 (AMR-WB), this is the same codec used for 'HD Voice', and I don't think it adds too much delay.
https://area-51.blog/2021/01/16/getting-a-dial-up-modem-work...
Call it IPOVOIP.
UK POTS is switching over to an IP based network, so does the bill specify the internet or ip based network?
If you submit a GDPR to your local constabulary, every phone call in the UK has a unique call identify much like a GUID now. Its quite insightful seeing just how much info is given out regarding the phone system capabilities via GDPR requests.