Update June 25 Latest thoughts here. June 24, Note, an update to this entry is posted here. July 2, Note an even-more current update to this blog entry is posted here. I would suggest reading it first. Update June 25, I have yet another followup here.
Several events before and during the AES convention last month here in NYC indicated for me a future for the world of audio networking for the live sound industry. I first really became aware of the IEEE Seeing working devices on the Infocomm floor inI thought this cool new technology might come on the market soon afterwards. I was wrong. The problem this causes is that AVB requires those special AVB-capable switches to function properly, so without a wide array of these on the market the standard is effectively stalled out from the user's perspective.
And worse, the rumor I heard from several people at AES was that Cisco--the pound gorilla of the networking world--was not going to implement AVB in their switches any time soon, despite being a founding member of the AVNU alliance. And around the time of my IOL visit, an article came out from Audinate talking about the situation linked from my writeup herewhich basically said that while they supported AVB, they—and the pro sound market—were not able to wait forever for affordable AVB-capable switches.
They also speculated that AVB switches didn't offer compelling value to our market, at least in the near term with high costs and limited availability. He is a very smart guy who understands way more about the technology than I would expect of the director of a company, and he clearly gets the market and the role of standards in industry. Ellison was clear in the talk and at lunch afterwards: Audinate plans to continue support for AVB. But Mr. Ellison also talked a lot about a new and potentially game changing standard: the then just ratified AES AES standard for audio applications of networks - High-performance streaming audio-over-IP interoperability.
Back at an Infocomm AVNU alliance cocktail party inI ran into my old friend Kevin Grossthe primary inventor and developer of Cobranet, which until Dante came on the scene was the most widely used networking standard for transporting audio in the pro Audio market.
Kevin guided the process to completion in about two years, which is an incredibly speedy time frame for the standards world but what the market needs. Dante and several other network systems operate at Layer 3. And this is great news for us today, since peering into my crystal ball, this is how I see the live sound networking world in the near future:. All manufacturers listed in the graphic have, to the best of my knowledge, working, actual product on the market using those networking technologies as of this writing.
As you can see, the scales are tilted pretty far over towards the Dante side, with Meyer and a few other key live sound players on the AVB side.Jun 6, Product announcements. It defines a protocol for messages that devices can send to each other in order to synchronise their clocks throughout a network.
The most recent publication of the standard achieves high accuracy timing to end devices through a combination of hardware time stamping of packets and accumulated delay calculations.
Accuracy can typically be achieved to sub-microsecond figures; this makes PTP extremely suitable for applications in audio and video networks. This revision improves the robustness, precision and accuracy, but it is not backwards compatible with the standard. PTPv2 brings more features to the timing fabric of real time networks. It is an advanced timing technology, which represents the future direction of accurate time synchronisation across all industries.
Since PTPv2 deals with timing and sync, many of the terms reference clocks of some kind.
Any device which contains a PTPv2 clock is itself simply considered a clock at the most basic level. The next distinction is between network and terminal devices. Network devices are generally considered to be pieces of equipment that make up the underlying fabric of the network: switches, routers etc. It is possible to find that various audio networking solutions each use different versions of PTP. Dante and Q-LAN used the original version 1, whilst Ravenna, which is widely used in the broadcast and studio world, uses version 2.
AES67, AVB/TSN, Dante Power AoIP’s Rise
With Dante systems, PTPv1 is used to synchronise clocks of all the audio device endpoints. This is an automatic process whereby a Grand Master is elected based on your preferred master settings for example and then the slave devices receive the clock sync messages.
In order to give PTP some priority on the network as it travels through the switches connecting your devicesDante uses Quality of Service QoS markings to designate PTP messages as having the highest priority. Clock precision can still be affected by the volume of traffic and how much contention there is for priority. Thus; PTP clock messages can get stuck and delayed in the backbone; in the switches between your devices.
As previously mentioned, v2 is not backwards compatible with v1. As more manufacturers have sought to bring AES67 interoperability and support to their products, they have harmonised their PTP implementations or included other mechanisms in order to adhere to the standard.
As shown in the illustration below, two clock domains two audio clock territories, Dante and AES67 will get created across the different devices.
In a PTPv2 network, one clock is elected as the Grandmaster. The Grandmaster is the clock from which all other clocks will derive their time; this is similar to a master clock that distributes Word clock in a digital audio setup. If no messages arrive after a predetermined length of time know as the Announce Timeout Interval the device acts as a master.
Announce messages also include information about the quality of the clock sending them. If a device receives an announce message and determines that its clock is more accurate, then it will assume the role of master and the other will revert to a slave.
This serves to ensure that the best clock on the segment is master as well as providing redundancy, in the case that a master clock goes offline due to failure. When other clocks notice the absence of a master, they will begin to announce their presence and the most accurate will assume the role of master. Figure 3 illustrates a simplified cycle of the BMCA. PTPv2 is capable of using hardware to solve some of the timing jitter encountered as packets move around the network and through different devices.
Devices can work out not only the time it has taken for messages to arrive but also update their time, or that of messages they send, to compensate for the delays experienced by packets as they are processed internally. This is also something which differentiates a Transparent Clock from a Boundary Clock. A switch acting as a Boundary Clock will use PTPv2 to update its own time and apply that to packets that it sends.
A Transparent Clock will calculate how long packets have spent inside of itself and add a correction for that to the packets as they leave. PTPv2 uses a series of messages to synchronise the time between clocks.AES67 is a standard to enable high-performance audio-over-IP streaming interoperability between the various IP based audio networking products currently available, based on existing standards such as Dante, Livewire, Q-LAN and Ravenna.
It is not a new technology but a bridging compliance mode common to all IP-Networks; an interoperability mode you can put an AES67 compliant device into, on any participating network. AES67 operates over standard layer 3 Ethernet networks and, as such, is routable and fully scalable, like any common modern IT network.
How can I educate my audio guys about AES67? Can I use AES67 for remote contribution? That will depend on the network connection between the remote and main site and the amount of latency it adds. What intercom system systems are available for audio over IP?
Since some AES67 implementations include provisions for interoperation with ACIP, they may also be able to interoperate with intercoms built according to Tech By buying AES67 compliant products you know that, whoever those products are made by, they will be compatible and thus you will able to stream audio between them. You should ensure the networked audio equipment you purchase supports AES67, or will in the future.
The app is currently available for iOS only and can be downloaded from the iTunes store:. I have a small home studio, what does AES67 mean to me? If you want to expand your studio and purchase AES67 enabled products, you will have the security of knowing that the equipment will stream audio to other AES67 enabled products you may want to buy in the future.Glimpse the Future of Audio Networking, AES67 & Dante Via - Aidan Williams at AVNW 2015
What is the value of AES67 to a Manufacturer? By implementing AES67 or having an AES67 compatibility mode you are not restricting to your networked streaming audio products working with just the streaming protocol you are using, your equipment is capable of streaming to any other AES67 product.
This allows your product to be used in a wider set of applications and environments, broadening its available market. Are there set up guides for AES67? In the fullness of time, the MNA hope to provide a series of documents surrounding setting up and using AES67 media networks.
Is there a Maximum Channel Count? The maximum channel count depends on the network infrastructure. As a general rule, a gigabit network should be able to handle channels of audio at 48kHz sample rate. Can AES67 streams be compressed to reduce the bandwidth consumed? No, the AES67 standard specifies that all audio is uncompressed. This is because AES67 is a high performance low latency streaming standard, compression will add latency to the system. How do I get AES67 into my product? Ravenna, Livewire, Dante.
Does the AES67 standard make it harder for manufacturers to provide unique advantages? AES67 is a standard for audio streaming and synchronization interoperability. AES67 was developed in recognition that new audio networking systems were using similar protocols and technology.
AES67 does not eliminate existing networking systems but allows manufactures to build bridges between systems by also implementing the AES67 interoperability mode. Manufacturers are also able to differentiate their products based on additional unique features, increased performance and the extent to which they support optional AES67 capabilities. Actual latency is determined by a number of factors, such as network environmental parameters i.
With a typical AES67 packet setup i.Update June 30, Latest thoughts here. Update June 25 Latest thoughts here. It's become an annual tradition-since I've been writing about audio networking from a live sound perspective after recovering from Infocomm. You can see last year's entry here. This year, it seems, there's not a whole lot to write, since there was more of the same, with some exciting directions for the future which I'll get to in a bit.
But the number of Dante products seems to keep expanding, both in the live sound and install markets. Here's the Audinate display on the Infocomm floor:. Audinate kicked off the week with their AV Networking Worldwhich as always features some interesting Dante-based products. Here's a couple favorites, based on Power over Ethernet my fave. PoE speakers from SoundTube:. They are also soon introducing Dante Domain Manager, which looks pretty cool for very large systems.
At the training as part of the event, Audinate rolled out Level 3 of their training program. Level 3 isn't really necessary for most users of Dante systems; it's really aimed at geeks working on large systems.
It covers a lot of foundation networking technology, and some very esoteric issues you might encounter in complex systems. I think it's very smart of Audinate to put forward this training; Levels 1 and 2 and some new tools available in Dante controller really changed the way I look at these systems, and increased my confidence when working with Dante networks, where so much of the underlying functionality is invisible.
There were of course some AVB products on the show, but in my world of live sound, I didn't see a whole lot of new development. There was talk of new AVB capable switches, but as of this writing there's still only one that's been certified Extremeeven though Cisco had an AVB switch on display last year. Looking on that certification list, I do see L'Acoustics has added a couple new AVB-connected amplifier products to their line ; unfortunately I missed their booth at the show.
I am excited to see that Meyer is now shipping their Galaxy Galileo replacement system I love the Galileos and use them a lotwhich interconnects via AVB:. As I wrote back inAES67an open standard audio exchange method, may be the key to the success of audio networking for the future and there was a lot of buzz about it at the show.
I still view this unified interface as a key part of the system; it's literally all that most users see of the system. And as of late last year AES67 is available in Yamaha products, and since we own a lot of Yamaha stuff at the school, I would love to get my hands on some AES67 compatible gear, and see what the process of patching and so on is like. If you have access to some shipping AES67 gear something like a stage box would be perfect please get in touch!
I posted a lot of Infocomm photos and updates on my twitter feed ; I have a photo gallery here as well. We also did our annual geekout; writeup here.Update your browser to view this website correctly. Update my browser now. By Steve Harvey. With live sound, post production and, increasingly, music production facilities now routinely utilizing audio-over-IP networks, looks like it could be the year that the broadcast world also reaches the proverbial tipping point and joins them.
Key to the proliferation of networking throughout the pro audio business is the AES67 high-performance streaming audio-over-IP interoperability standard, which has come to provide the common thread—the one ring to rule them all, if you will—between the many and various networking schemes available.
For example, AES67 has been adopted for audio transport by the SMPTE ST suite of standards for professional media over managed IP networks, critical to the broadcast industry, which is well on its way to being published in full. Yet the publication of a standard such as AES67 is only the start of the process. Initially published in SeptemberAES67 was released with an addendum inwas revised in and, on April 15,was revised yet again.
The PICS serves a variety of purposes. For instance, it can act as a checklist by the protocol implementer to reduce the risk of failure to conform to the standard through oversight, the AES statement notes. Alternately, it may serve as a detailed indication of the capabilities of the implementation relative to what is understood to be included in the standard PICS proforma.
Lastly, it can be used to check potential interoperation with other AEScompatible implementations. The AES has also established a task group, SCR, to define a standardized method for transporting metadata associated with audio in parallel to, rather than part of, the AES67 stream. Standards work requires funding, of course, and the AES has just acknowledged the support of three manufacturers in its AES Standards Sustainer program.
Waves has joined at the Silver level, as has PreSonus, and Attero Tech has joined at the basic level. In mid-April, the AES also announced that, in response to requests from manufacturers implementing the standard, it has introduced an AES67 logo for use on goods and literature to identify products that conform to the specification.
It promises that further details of the logo and its license terms are forthcoming. According to the latest announcement from the company, there are more than 1, commercially available products supporting Dante from plus AV manufacturers.
In May, the company announced that more than Dante-enabled products from various AV manufacturers now feature upgraded firmware supporting Dante Domain Manager, which adds such features as user authentication, role-based security and audit capabilities to its scalability and organization of Dante systems over a network topology. Want more information like this? Subscribe to our newsletter and get it delivered right to your inbox.
For more stories like this, and to keep up to date with all our market leading news, features and analysis, sign up to our newsletter here. Your browser is out-of-date! Subscribe For more stories like this, and to keep up to date with all our market leading news, features and analysis, sign up to our newsletter here. Related Articles.Updated: Mar 22, Audio Networking is becoming increasingly popular. A growing number of professional AV products not primarily designed for audio, including AV over IP solutions, do however not incorporate Dante support, they instead offer integration with DSP's and other 3rd party equipment through AES So what's the difference between these standards, and what are the implications to the integrator and the end users?
The problem is that these standards are mostly incompatible and non-free. If an AV manufacturer would like to implement audio streaming using these technologies, it typically involves licensing fees, which in the end increases the cost of the product to the end user.
There is a possible solution though; AES67 enables all of these different technologies work together, and it's free! Too good to be true? As with most things free it comes with a few limitations. AES67 is an interoperability standard for professional, low latency audio over local area IP networks.
It was designed around existing protocols, which means it shares characteristics with both Dante, RAVENNA as well as other audio streaming technologies. Existing technology providers who wanted to support AES67 were free to either implement interoperability as a special mode, or transition to use the features specified in the AES67 standard as the native mode.
AES67 was never meant to replace Dante, or any other existing technologies for audio transmission. It was designed to act as a bridge, allowing communications between systems designed on different technologies. The latest version of the standard is AES The standard defines:. One of the features missing on this list is discovery, and there seem to be a bit of confusion on what this means. The challenge is that the specification offers multiple alternatives, including Bonjour, SAP and others.
Lately SAP does seem to be the more popular choice though, largely helped by Dante's market share. AES67 also lacks control and diagnostics capabilities. Dante offers some very useful tools for monitoring clock accuracy and latency, and pin code protection for security is also not available for AES The big question is if what we can expect from audio quality and robustness when AES67 is integrated in a Dante system.
After reaching out to Audinate, they did confirm that their current end-to-end latency in AES67 mode is fixed at 2 times the packet time; 2 ms, but will in fact be increased to 3 ms with firmware 4. AES67 integrated in a Dante system is also limited to multicast only, and 24 bit audio at 48 kHz sampling rate. Other compatibility concerns worth mentioning are the issues with synchronization and QoS. Audinate has later added PTPv2 support to Dante to ensure compatibility, but if you are using an external clock, make sure this device supports PTPv2.
There are also conflicts related to QoS, but this is also manageable. Integrating devices with AES67 audio into a Dante application comes with a few limitations. The key issues are a fixed latency and audio settings, lack of monitoring and diagnostics tools, and possibly having to use additional software to manage discovery if the devices aren't using SAP.
What are the benefits of AES67?
Weather these issues are manageable or not, largely depends on the application. Regardless, AES67 is a huge step forward in integrating different audio technologies in the same network.While AES67 does not provide performance improvements beyond what Dante already delivers today, the inclusion of AES67 in the Dante solution enables interoperability with other AES67 implementations by other compliant vendors.
Today, Dante provides interoperable audio networking between hundreds of products developed by our licensees. To achieve interoperability, AES67 mandates a specific RTP payload format for delivering audio over IP networks, as well as methods for exchanging information about audio streams. RTP is already used extensively in communication and entertainment systems that involve streaming media, such as VoIP telephony, video conferencing, and IP television.
AES67 offers the potential for lower cost network transport built on mature standards when compared to other less widely adopted industry standards. AES67 can exploit Ethernet switches supporting the IEEE precision time protocol and Quality of Service QoS but, unlike some other network standards, does not depend on specialized switches in order to operate. This allows for audio over IP solutions to scale beyond simple local area networks, passing through routers as well as switches. This potentially opens broad new markets for audio over IP solutions.