JackTrip: Multi-machine audio network performance over the Internet | Hacker News Reader

JackTrip: Multi-machine audio network performance over the Internet

github.com

53 pointswallflower18 comments

I've found SonoBus easier to work with, thanks to its availability as a VST plugin and its Android and iOS apps. Easy to connect to the netjack2 system in my studio.

May I ask how you use it as a VST plugin, what makes the apps nice (is it only that they are available?), and what and how you connect with netjack2?

Loading the VST in Reaper makes it fantastically easy to turn just about any Android or iOS device into a monitor. Great for checking the final mixdown for YouTube videos and EQing my podcasts in real time. It is also handy for bringing a guest onto a show since I can wire things up to use the mix minus in the studio without relying on AEC.

I did my best to explain the NetJack2 layout in my studio in this post https://interfacinglinux.com/2024/03/02/lossless-network-aud...

Looks seamless when you use it like that. As a "normal user" of a desktop OS, network audio always seems so custom, I guess how the OS and applications handle audio is always going to be the bottle neck. I have no need for a DAW but I often want to stream audio from something to something else e.g. as you say monitor audio on some other device.

Side note: SonoBus internally uses my AOO C/C++ library for the actual audio streaming and peer-to-peer communication. See https://aoo.iem.at for more information.

The Reaper DAW can do this too: https://news.ycombinator.com/item?id=31018600

For somebody not in this space is anybody able to give me a simple explanation and use case for this? The README wasn’t so enlightening.

https://jacktrip.github.io/jacktrip/

https://stoor.net/pages/about

STOOR did a live multi-artist techno performance between two studios in Netherlands and Germany.

They might have used something like JackTrip to synchronise their instruments.

While I'm definitely a fan of Open Source, I wonder how this compares to Dante, and if you could get Jack and Dante to interoperate.

Easy, they don't really compare. Dante is meant mostly for on premise, low latency audio transmition between audio equipment that is made by manufacturers willing to pay the fee. A normal Dante Network that likes to keep those low latency guarantees will not leave the premise.

This looks more like it is meant to do audio transmission over the internet between computers.

JackTrip can be used over the internet. Dante is LAN-only.

Dante is proprietary. As I understand it, it doesn't interoperate with anything (without incorporating licensed Dante IP in your product).

The open standards in a similar space (along various axis) as Dante are RTP/RTSP, AES50, AES67, AVB, AVB/MILAN. Often on dedicated networks. JackTrip is closest to RTP: lossless audio in UDP packets. Best-effort clock-recovery based on packet arrival times.

The main distinguishing features of all of these protocols are: - Ethernet Layer 2, or 3? - Packet formats - Method of time and clock distribution/clock recovery (PTP1 or 2, other) - Control plane protocols and management interface - Dependence on PTP and traffic reservation implementation on all intermediate Ethernet switches (AVB, Milan: yes, others: no)

MILAN (an AVB profile/subset) is the most direct competitor to Dante (aiming at the Plug-and-play capabilities of Dante). None the less Dante is actually a product offering, whereas MILAN is a technical standard.

From a technical standpoint clock distribution and control plane are the big differentiators.

If anyone wants to talk about open source development in this space please get in touch.

Hey! maybe unrelated but I want to acknowledge your work in the audio field. I was myself involved in the development of audio technologies for the hard of hearing for over a decade, including a low-latency audio transmission protocol over WiFi. In my junior days, I learned a lot about audio thanks to your blog. I remember your "Real-time audio programming 101: time waits for nothing" as one of the most influential articles that shaped my understanding of real time audio. Thanks so much for sharing all that knowledge!

No problems, I'm glad you got something out of it. I learnt most of it from asking dumb questions on the musicdsp mailing list back in the day.

Just gonna throw in that the now standard Linux audio daemon PipeWire has aes67 support:

> PipeWire builds pipewire-aes67 binary which has RTP receiving and transmitting modules configured for communicating in professional AES67 networks. It has been tested to work with Dante and RAVENNA networks - https://gitlab.freedesktop.org/pipewire/pipewire/-/wikis/AES...

A very sizable portion of the page is about setting up PTP. But it doesn't mention setting up PTP on intermediary switches, which is new to me!

I'm not 100% clear on the exact PTP requirements of each standard. I'm pretty sure that Dante uses PTPv1 and AVB uses PTPv2, and very confident that AVB requires all switches to be running PTP.

What's the latency involved?

They advertise 25ms https://www.jacktrip.com/

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