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Comment by reaperducer | Hacker News Reader

I had the pleasure of using a real landline just before the pandemic. Honest wire-to-wire connection between two ranches, so no silly VOIP steps between.

It was fantastic.

You don't appreciate how much latency is destroying our ability to communicate verbally until you go back to the old way.

One example is arguing. It's no wonder people used to be able to argue with one another on a telephone. You could raise your voice and still hear the other side and adjust your speech in real time. Today it's just one party shouting over the other to drown the opponent out.

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Between miserable latency, not-so-great fidelity, and the fecklessness of phone companies in the face of the robocall epidemic, I have come to hate phone calls.

I'm rooting for something to replace phone communications. Any chance that Matrix can do better on any of those fronts? Especially on fidelity and latency since they're germane to the high-level subject of this discussion.

One conspiracy theory is that tech companies have lobbied to prevent real action on robo-calls, in order to get people like you to hate calls and migrate to online services.

> conspiracy theory

Sounds like a run-the-mill everyday business decision.

I don't think the FCC needed any lobbying to do nothing for the consumer these last few years.

That seems unlikely. The robocall phenomena don't really exists in France and many people still dislike calling

It definitely exists in France. I still receive a robo call every single day on my land-line, liste rouge/Robinson or not. I have stopped bothering reporting them. The prevalence of "who called me/is it important" services makes me think it is not an uncommon problem.

Another thing I clearly experienced in the beginning of the mobile phone era, when one still could compare those things, was dynamic range. When speaking via a good landline, you could hear the presence of the other, but on mobile, there was only speech or silence. Noise suppression saves bandwidth, but it is so aggressive that the nuances in the other's utterances (breathing, hesitations, etc.) simply disappear. For a simple transaction that's not a problem, but when your SO lives far away, miscommunication arises too easily.

VoIP is definitely not the issue here. Codecs can be, and are, fast. I have no idea what cellular providers are doing to mangle the voice path so bad, but it certainly isn’t inherent to VoIP.

I thought the delay would be because VoIP uses packet-based communications rather than a direct connection.

Most landline networks are also digital now for exchange to exchange communications, so even those will be packet based

Not all packets are created equal. ATM (Asynchronous Transfer Mode) cells have a fixed 48-byte payload, but IP (Internet Protocol) packets have longer headers and variable-length payloads up to about 1500 bytes. ATM is designed for low latency audio transfer, whereas IP incentivizes using bigger packets to reduce overhead at the expense of more latency.

Historically, sure. But at a realistic modern internet speed of, say, 15 MB/s, 1500 bytes takes 0.1ms to transfer. Even at 1.5 MB/s it takes 1ms. Typical VoIP call latency apparently ranges from 20ms to 200ms, so packet size is not a major contributor.

I have no idea what they are doing but it sucks. I have in many occasions found that making a call using WhatsApp works much better - both in quality and latency - than making a phone call. Phone lines are so atrociously bad, they've put me off from doing a voice call unless it's absolutely necessary.

Ever experienced european ISDN to european ISDN at 64kb/s ?

I know americans said 'I still don't need(it)' but I still do miss it :-)

Funny thing was I had better(and cheaper!) calls to the US using calling cards, dialing into Frankfurt, and from there to the US than using the native offer of my telco.

Arguing is definitely affected by latency but it is at least possible to reduce latency, e.g. by getting fibre internet or using ethernet.

A more annoying "feature" of many VoIP systems is that they mute the other person while you're talking. You literally can't interrupt people because they won't hear you.

I presume this is done in order to reduce feedback, but it still sucks.

I'm glad you bring up latency - I've experienced several second latency in discord and it's really terrible when it happens (diagnosed via side-channel). The worst thing is that the app does nothing to try to salvage the conversation and the app doesn't tell you it's in a degraded state, so you'd never know this was a problem.

It has a quality indicator on the bottom right, and if you click it there's a nice chart that shows the latency.

https://i.imgur.com/vR7NSpG.png

Or are you talking about something else?

Well, I stand corrected. I'll take a look at that chart next time I experience something like that happening. I haven't seen this UI before. Thanks!

Add to that when the other party is hearing you over speakers, and the echo cancellation kicks in whenever you start to speak so that you can't hear what they're saying when they try to interject. You just see lips moving, stop speaking and you start to hear them mid sentence.

I remember switching to VoIP and noticing the voice quality getting better compared to the old twisted pair. No noticeable latency.

A cell phone tbh, is about the same latency as a landline in most end to end call circumstances. Latency only really is noticeable when better than 600ms. (And only a real problem over 1000)

I think latency is noticeable at even lower values. As a basic example, try to sing a song with someone over a voice call. Consider using Airpods or similar bluetooth headphones to make it more apparent.

I disagree with these numbers, in general. Though of course "noticeable" is subjective and varies by use case as well as by person.

For many people, end-to-end audio latency in a 1:1 conversation becomes noticeable/annoying at 200ms. And in a multi-participant conversation, talking over each other becomes noticeably more common even at 100ms compared to 50ms.

IIRC the Bell standard points out no more than 50ms.

Imagine trying to have a face-to-face conversation with 600ms of latency...

There is a huge quantity of research opposing your statements. The ITU considers 300ms round-trip latency to be catastrophic.

Do you have a citation for that? I've heard 200ms as the key threshold and the ITU uses 100ms for their default delay sensitivity class[1]. One key concept is that this isn't fixed but situational: if you're watching a TV show, the threshold is higher than if you're trying to react to something which is higher than simply noticing a delay and speech is more forgiving than, say, music hitting a precise tempo (I believe musicians have been tested as noticing delays down into the 10-20ms range).

One other big factor is consistency: if the variability is due to compressor overhead which is constant, the effect will be noticeable but less distracting than if it's varying due to something like wireless conditions.

1. https://www.itu.int/rec/T-REC-G.107-201506-I/en

I know that if the delay is constant, a good musician can compensate for it. I think it's a lot easier to hear that two things don't happen at exactly the same time than it is to tell whether two things happen 200ms or 300ms apart.

Musicians have that internal metronome to compare things with.

Do you have a source for that? Gaming latency is noticeable in the ~100ms range, and humans are highly sensitive to small adjustments in speech.

I would've thought it would be closer to the 200ms range, but I don't have any data to support that.

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