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So what?

Do you mean it is not analog and latency is higher as a result? Then yes, it matters. I hate latency in voice calls, I already went into arguments because of that.

I remember in a remote work meeting, we had a frantic discussion, with some disagreements and strong opinions, but it was productive and purely technical, nothing personal. But then someone angrily told me "stop interrupting me!", the thing is, I wasn't, and then, I realized that the latency was messing with us. Because of the latency, from her point of view, I interrupted her, and from mine, she interrupted me. That's when I realized how much it mattered, we simply can't have a normal conversation with high latency. Either we deliberately take turns, as if it was a traditional 2-way radio communication, or we may get these awkward situations, neither feel natural.

High latency can be as little as 100ms (corresponding to about 30m of distance in real life).






It still bothers me. Analog and TDM voice was magical and we didn't appreciate it until it was gone. VoIP was so much cheaper, the latency became the norm, and people who've never known anything different simply have no idea what was lost.

It used to be that if you had two landlines in a large room, you could call one from the other, and your voice would go into one phone, electrically go across town into the switch, back out the other line, and out the other phone, before the soundwaves traveled the length of the room. It was _so_ good.


> interrupting [... timing ...] turns

There's an old linguistics tale, AFAIR fuzzily, of Inuit kids going off to boarding school, and upon return, having lost interest in hearing from adults. The kids believe the adults have little to say to them. ... Because the kids' conversational turn-taking invitation pauses had shortened, and were going unrecognized.

> as if it was a traditional 2-way radio communication

If there was a low-latency side-channel for the end-points to coordinate, they might provide mic clicks and carrier noise for awareness? Like an electric car playing engine rumble.


> Do you mean it is not analog and latency is higher as a result?

Digital teleophony doesn't imply significant latency. PRI calling (T1/ISDN) is digital, but the sampling delay is minimal, and it's sent one sample at a time, so there's no packetization delay.

VoIP tends to run a codec with sampling/encoding delays, and tends to be at least 20ms packetization, and then you have a jitter buffer and probably input and output buffering too.


The main problem is packet switching instead of circuit switching. Internet sucks for voice

Almost all my calls these days are via OTT VoIP (FaceTime, WhatsApp, Zoom etc.), and I really can't say I notice much of a qualitative difference to old PCM lines – if anything, the opposite is the case; wideband voice makes longer phone calls much more bearable.

And that's over unprioritized 5G/LTE and Wi-Fi. Properly prioritized VoIP, such as the one used in VoLTE and NGN networks, should have even better jitter and latency characteristics.


I mean.... when was the last time you made a call that was full analog/PRI? Did you notice that cell phones had higher latency than landlines?

30-40 ms latency is probably the lower bound on a VoIP call [1]; most calls will be a lot more. But you might not notice it if that's all you have access to, or if all of your calls are long distance that it would never be good anyway. On long distance, you probably have a better transmission delay now over internet than you would have had over telephone networks, as cable routes have improved and routing is often more direct; that will probably offset some of the packetization delay.

[1] You can do better, but it involves having great connectivity and sending lots of very small packets, and mainstream calling isn't willing to send 200 packets per second of 5 ms of samples when most people don't notice/complain about the latency when sending 50 packets per second with 20 ms of samples.


> when was the last time you made a call that was full analog/PRI?

Probably 15-20 years ago? That would have most likely been over DECT, though, which adds 10 or so ms by itself (our wired landline phone was in a slightly awkward location :)

> Did you notice that cell phones had higher latency than landlines?

Compared to landlines, the most noticeable aspect of GSM wasn't the latency (unless the person on the other line was right next to me, in which case there was an echo), but rather the absolute potato quality compared to both G.711 landlines and modern VoIP codecs.

> mainstream calling isn't willing to send 200 packets per second of 5 ms of samples

Yeah, that would probably be too much overhead for most applications. But now you got me wondering: Do video calls have lower latency audio (assuming the codec can do better than 20ms in the first place), given that there's probably much more data available to send at any given time?


> Do video calls have lower latency audio (assuming the codec can do better than 20ms in the first place), given that there's probably much more data available to send at any given time?

At least for WebRTC, no. The audio and video streams are separate, there's no mechanism in WebRTC to piggyback audio onto video packets. If the receiver is synchronizing audio with video, there's potential for additional delay (but when A/V sync works, it's probably worth it)

I don't know details about WhatsApp calling (although I worked there, I didn't touch the realtime calling stack), I think it does use RTP which isn't really built around piggybacking, but since it's a closed service, they can do whatever. No idea about Apple's calling either.


Ye latency is a major headache. You can't gave normal conversations. I think it is mainly a property of overloaded servers and background noise cancelling etc.

I didn't have these problens when running Ventrilo or old p2p Skype.


> So what?

It's not a microphone in a reciever oscillating a copper line. In my opinion, if it's voip it's not a "landline".


For me, if it is a line and it goes through the land (i.e. by wire/fiber), then it is a landline. As opposed to signals going through the air, i.e. by radio.

Except in very special cases, like emergency phones in ships, it is never a direct connection, there are at least amplifiers in the middle. So between a direct connection from the microphone to the speaker at the other end and VoIP, where do you draw the line? (pun not intended)


There almost certainly still is one – from the handset or headset to the VoIP desk phone :)



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