> Latency will always be an issue here.

No, it will not. :) You can get to 5ms above the network delay, even using compressed audio (opus), though some sound devices may be picky.

Network delay can be just a few milliseconds even across a city, assuming that no latency murdering devices (wifi or nics in interrupt mitigation mode) are on the path.

This means that you can have lower audio delay from a compressed audio conference crossing your state than is achieved from speed-of-sound delays from a performer sitting on the other side of a moderate sized room!

The future is already here -- it's just not very evenly distributed. (yet)

> Network delay can be just a few milliseconds even across a city

If you limit your collaboration radius in that way, then sure (though it still depends on the router technology the packets encounter along the way). But if you want to go beyond that, the numbers get bigger. For transcontinental stuff, much much longer.

But thanks for correcting the basic thrust.

One way delay between San Francisco and New York city is 37ms across the public internet today.

This is the same one way delay as sound in air across 12 meters, which would be on the large size for an orchestra pit.

It does get larger as you go further, but live performance with people on the other side of a large continent is completely realistic and has been done. :)

Sure, expecting to go between NYC and someone in china behind the GFW is probably asking too much for a seamless experience. But many people mostly want to work with people in the same country as them ... and for that, with sufficient technology, latency need not be an issue.

> live performance with people on the other side of a large continent is completely realistic and has been done. :)

In case you're not aware, I'm the original author of JACK, around which JackTrip is built, which is the most likely and reliable tool for such a collaboration. I certainly know people who've done this, and I regret that I didn't realize back in February how useful it would have been to make something like JackTrip into a much easier-to-use tool for computer-naive folks.

37ms cross-country in the USA is optimistic, but certainly possible. US->Europe is not so great.

In case you're not aware, I'm one of the authors of Opus. :P

When the pandemic hit I thought about putting out some pointers to easier to use low latency streaming resources... but... that would require overcoming pandemic lockdown funk. And honestly it's still kind of a technical rats nest to get everything working. It's not exactly musician friendly to need instructions like "next you need to make sure your nic doesn't enable interrupt mitigation when there are more than 100 packets per second...". :)

[I love Jack BTW.]

Another example of HN being amazing. In the interest of playing too, I did some work on reducing audio latency in Android, but that all predated AAudio/Oboe, which takes that quite a bit further and makes at least the Pixel devices a plausible platform for low-latency audio collaboration apps as suggested in this thread. I for one would love to see that happen - I miss singing hymns on Sunday mornings.

Yeah, I'm really sad that I didn't get some serious effort together on this back at the start of the year. There's no lockdown funk here, just too busy working on Ardour.

One of the 2 netjack implementations uses Opus - great stuff.

Absolutely right that the good stuff (i.e. not zoom) requires way too much setup, and even then (as you note) you're not guaranteed reliable function because you don't control all the hops. Soundjack gets the user-side fairly good, but it's still not quite what I think would have really taken off during the pandemic - probably needs a mobile app for that.

I love libopus! I use it as one of the codecs for my low latency streaming library “AOO“ https://git.iem.at/cm/aoo

And I love AOO, I use it as the basis for SonoBus.... I’ve been meaning to contact you about it, and work on merging your latest into my fork (which has diverged a bit over the last few months).

And hey Paul, long time no chat!

https://sonobus.net

Thank you for JACK btw.

Latency in this case is cumulative. 1-5ms for DAW encoding of your analog signal, encoding it in whatever codec is used for transmission 1-?ms. Actual transmission 10-1000ms. Decoding the signal into audio channels for playback 1-5ms. Don’t forget this is duplex. You are transmitting as you are also receiving. In an ideal world we could get very close to zero-latency (enough where an orchestra could by in time step with each other).

It sounds do-able. It sounds like we should already be there. Looking around though you’ll see we aren’t because of technical issues described that are systemic of the infrastructure used.

JackTrip doesn't encode at all, it sends raw PCM over the wire. So that helps.

Network delay can be just a few milliseconds across a city, or you can discover, like my brother who teaches elementary music, that your kids have different ISPs and some are awful.

He's been teaching music remotely since August, and one of the first days he demonstrated why they have to be on mute on Zoom by having them all sing happy birthday to one of the students. It was utter chaos. I worked with him to try to find a solution, we tried various tools, but found nothing that worked across cable, DSL, WISP, and cellular connections.

Some tools could sync after the fact so the recording sounded good (if each musician keeps the same tempo), but the whole point is that the kids needed to be able to hear and follow each other, and just one student at 500ms latency totally ruins the tech.

So he teaches music to second graders by zoom, and they're all on mute unless he unmutes them one by one.

> Latency will always be an issue here

Yes, it will as soon as you are 1000km away. Light speed creates a lot of unresolvable latency. How can you play live between London and Sydney? You need a different musical understanding.

1000km at the speed of light is 3.33 ms which on it’s own isn’t an issue. While fiber is slower, you can have low latency connections over that distance without issue.

The real problem is computer audio isn’t designed for low latency. Generally, as long as video and audio are reasonably synchronized nobody notices, but latency as rarely a consideration on it’s own.

This shows actual network latency between regions.

https://www.cloudping.co/grid

Even with the state of the art equipment, transcontinental music making will always exclude the kinds of music that most of us play while in the same room together.

I was specifically mentioning 1,000km as the metric.

AWS latency isn’t great between regions. Looking on some other sites you can see London to NYC is 70ms, but that’s also 5,500km which is far behind the 1,000km mentioned. Ex: https://wondernetwork.com/pings/London

70ms /5.5 = 12.7 ms seconds round trip. The speed of sound is only 1,125 ft/s, and half that for round trips so 1,000km over fiber is the equivalent of ~7.14 feet apart in the same room plus whatever latency your personal computer and software adds.

My DSL line gives me a 20 ms roundtrip to the first hop. My only choices at this address are DSL, LTE (latency seems highly variable, and much more expensive than DSL), or paying at least $50k to get muni fiber installed. So latency will always be an issue here.

Well, I could probably get some form of POTS either analog or ISDN/T1

DSL is pretty much the worst of the wireline services as far as latency goes. Even some of the wireless technologies are better.

If you use sonic, at least in some areas they have a setting where you can turn down the level of forward error correction and get lower latency in exchange for somewhat higher packet loss.

>Network delay can be just a few milliseconds even across a city

Sure. If you constraint it a little more, you can also just play to the persons in the next room directly (no streaming required).

For country level or worse global collaboration latency will always be a problem however...

Jitter is an issue as well. If the packets get routed in different ways the variability in latency might annoy you. Ever had a video call where the person gets really slow and then speeds up again?

For extremely low latency audio, you simply convert most jitter into loss (and the rest into added delay). How annoying the loss depends on the quality of the loss concealment.

If your conferencing is 'slowing down' and 'speeding up' then it's simply not a realtime communications medium.

> Network delay can be just a few milliseconds even across a city, assuming that no latency murdering devices (wifi or nics in interrupt mitigation mode) are on the path.

Even with fiber all ISPs I have ever used at home have latency murdering devices between my modem and their connection to the Internet. And I have to go through these hops even if I connect to an IP on the same IP.

Best I ever got was 12ms. Once you connect your laptop to the Internet in a data center you will instantly know what I mean.

Using Wi-Fi in my office adds up to 150ms of latency. (20m cat5 Ethernet cable has < 1ms)

Is that with bad signal? Sounds like something is wrong with your setup.

Got a Unifi AC LR and it only adds about 4 to t 6ms over Ethernet.

High jitter also occurs when the signal is marginal and the client scans the WiFi frequencies for other base stations to associate to (on the assumption that maybe you moved and there is another base station with a stronger signal). Since it has to “tune” into the other channels to listen for beacons, there will be a noticeable jitter while it is not tuned to the “current” channel.

This is my understanding as I’m still learning. Fascinating this stuff works at all.

Not bad signal, but the basis of how wifi shares the spectrum is time distribution.

Mumble is in fact one of those audio-only options which can do 20-50ms latency, which is why the parent suggested it.

I use Jam Kazam weekly and with a non Wi-Fi connection I get really good results

I mentioned JamKazam in this thread above about latency. It’s the top of my list for this purpose but still has some issues when the connection quality isn’t good (ISP issues)