AES67 resources – Audio over IP protocol
hartung.io
hartung.io
Nice thing about AES67 is that you can also, in theory, run the preamps and ADCs off PoE.
A lot of the time, we're needing to connect devices to Dante through a Dante interface (which can be expensive) or via a Dante Virtual Soundcard.
It will be interesting to see AES67 appearing more in consoles, especially lower end consoles. Hopefully this will open up Audio over IP to more audio professionals without resorting to proprietary protocols which can make it hard to connect systems from various vendors.
But isn't this dead before it really lifts off? Intel 1588 is not supported by every customer-NIC, and far from supported by most switches and such.
So if i have to build a "special" hardware environment anyway, why use AES67 and not some other proprietary solution without IP which may (? or may not) deliver better latency and jitter?
Last time i checked (which is a while back), the same support-argument would apply to multicast.
Am i missing the point here? Please enlighten me. Thank you!
This might be best explained by analogy: VoIP. Various issues including security model, DHCP-based provisioning, and QoS mean that corporate VoIP phones are typically deployed on a specially prepared network using a dedicated VLAN. While various VoIP vendors claim that you can just drop their solution on your existing network and have it work, this rarely pans out, and it's very common for introduction of VoIP to involve upgrades to network appliances and possibly architectural changes.
It might also be important context that prior to the adoption of VoIP most corporate environments were already using a proprietary digital telephone system such as Nortel Meridian, so it's not even a matter of upgrade from analog to digital.
Rather, the IP-based network, even with special requirements, is more flexible (due to the large set of routing, switching, etc protocols available for IP) and less costly to maintain (due to common skillset and equipment with computer networks). Even better, while there may be a new network investment required to switch to VoIP, that investment is "dual purpose" and the new network equipment will also serve your computers.
I'm not an expert in this field, my experience being limited to some work with Dante equipment years ago---VoIP is more my wheelhouse. But I think the situation is very similar here. Adopters of live audio over IP will almost certainly need to invest in new network equipment and possibly rearchitect their existing network. VLAN segregation and special QoS policy will presumably be the norm.
But choice of vendors and common skillset, if nothing else, will make the IP network less costly to maintain. There's a wide variety of technology available for moving IP around in interesting situations, fiber is popular in large theater contexts because of a perceived improvement in reliability over long runs (probably less significant in the days of GbE but I haven't been involved in this kind of thing in a while). Further, with sufficient precautions in place the network can be dual-use and can also serve purposes like administrative networks and even front-of-house wifi.
Protocols like Dante are already being widely adopted, and compatibility can be a big headache, so I think a uniform standard for this kind of thing will be very popular.
But is a latency of 2ms up to 50ms really attractive? I am in no way near audio engineering, but i can remember the MIDI-folks swearing about their 2ms latency.
The problem AES67 solves is that there are multiple competing standards and it's difficult or impossible to connect them over the same network. The standard is there to unify under one protocol. The tech problem was delivering audio in sync over large networks, and rather than use custom hardware they used commodity networking gear and network protocols (not doing that might seem crazy, but that's generally what happens in high end audio). But even then, you probably have some special hardware in your environment to connect microphones and speakers to the network in the first place.
Not for consumer products, but it's not optimized for those. If you buy any kind of professional NICs and switches, 1588 support isn't that hard to find. Wide support means less trouble if one manufacturer changes things, more choice in hardware than a proprietary system would likely have. Can live with existing networks.
E.g. lets say you're building out an event at a large convention center. They probably can give you Ethernet or fiber from one end of the building to the other, but can they give it to you for your proprietary thing of choice?
Or you are making a permanent installation: do you want to have to put one vendors stuff everywhere, as a parallel network of proprietary stuff, or do you prefer something that works on the network infrastructure you already have, with maybe some upgrades to some components, but keeping it with the stuff your network people already know? Especially if it means you don't need to pull additional cabling or fiber everywhere?
No, because automotive is the use case driving this and consumer/professional audio will come along for the ride. Think about how you synchronize the LCDs and speakers in an SUV--Ethernet is way easier than just about any other solution.
The biggest obstacle to adoption in the consumer/prosumer space is the fact that everybody killed Ethernet ports on their laptops.
This isn't how you connect something trivial like an LCD to your speakers, it's how you rig up the audio system at a theme park or network the audio feeds for all the broadcasters at a World Cup match. Those are two applications that AES67 committee members had worked on and where it will probably be deployed.
If I'm being snarky, AES67 is a last ditch effort by the proprietary vendors to remain relevant before AVB/TSN wipes them out.
Before Covid, it seemed like Presonus was wiring up every new church I knew of with AVB/TSN.
> This isn't how you connect something trivial like an LCD to your speakers
Automotive companies do not regard that as trivial. Copper is heavy and expensive and difficult to route. LCDs are in the ceiling and speakers are in the floor. The only common point is at the (literal, in this case) firewall.
Collapsing everything to a few pairs of Ethernet is a big deal for them.
My personal opinion is that AES67 will eventually win. Dante has the installed base for now, and it can operate in an AES67 interoperability mode (albeit with quite a few limitations). SMPTE 2110-30 is essentially AES67. AES/TSN will limp along within homogeneous networks with Dante/AES67 at the edges where interoperability is required.
Don't get me wrong, I like AVB (see other post about AES67/AVB bridge), even though it has some quirks, e.g. the relationship between PTP and media clocks is a lot more straightforward in AES67.
Onsite, it's not feasible to use miles of cable for analog signals (and it is miles, in large venues). Even with balanced connections you start getting noise problems after a few hundred feet. Digital conversion closer to the sources solves this issue, which means those DACs and ADCs need to be networked somehow. Building infrastructure for corporate networks onsite is a mostly solved problem with lots of cheap hardware available, so they just use that and place custom gear with the converters at the input/output locations. Not to mention if lighting is involved, noise can get really quite bad.
It's just all around more modular, cheaper at scale, effective, and foolproof than full analog.
There's also a whole bunch of less sexy use cases in corporate environments where a PA doesn't work at all.
I’m not in the industry but my wife is, and AES67 seems like a complete game changer.
And that was just the early days, when even the protocols that sometimes used Ethernet were only L1 and L2 and not IP-compatible/routable. You might have had several lines of ethernet running back and forth to the stage, but couldn't comingle snakes with IP-based control systems.
AES67 and friends pushed the industry to a point where, basically, everything speaks IP. The stage remains an analog realm up to the DIs and mic premaps, and everything can be routed and distributed in almost infinite combinations from there with commodity hardware.
Its seriously as impactful as the shift from tape to digital recording in terms of the new workflow options and paradigms it opened up.
Why is a standards group paywalled!
This is their business model. You pay for access to the standards documents, and that money funds the development and maintenance process. It's supported by governments requiring adherence to these standards, so implementors are obliged to purchase them.
Gratis access to standards is a newer model which relies on a different funding stream. The IETF is one such example.
Certifiable Precision Timeing over anything is not trivial.
The Audio Engineer Society has been around since 1948 (https://en.wikipedia.org/wiki/Audio_Engineering_Society).
It released the AES/EBU standard (https://en.wikipedia.org/wiki/AES3) for digital audio I/O in 1985, and AES/EBU is still used on some pro audio equipment. So in a digital audio context, "AES" is already familiar.