Understanding the Livewire+ AES67 Protocol
Over the last 15 to 20 years, broadcast audio systems have undergone revolutionary changes—from analog to digital technologies, from manual to computer-assisted workflows, integrating related systems such as telephones and intercom systems, and much more.
One key enabling technology is Audio over IP (AoIP), which delivers a number of significant benefits, including operational flexibility, scalability and lower costs. However, implementing AoIP systems to take advantage of these benefits depends almost entirely on the existence of and support for interface and interoperability standards to ensure that the many elements that make up professional audio systems are capable of working together. To that end, there are several AoIP protocols designed to achieve this goal by easing implementation and integration, including the increasingly popular Livewire+ AES67.
Originally known as Livewire, this pioneering technology was created in 2003 by the Telos Alliance as a means of transmitting low-latency, high-reliability audio over switched Ethernet. Livewire+ AES67 adds full compliance with the AES67-2013 interoperability standard for high-performance AoIP transport over IP audio networking products. This allows devices to seamlessly connect directly to Livewire+ networks for connecting audio streams, regardless of device type or manufacturer. Livewire+ AES67 also offers the flexibility to incorporate and comply with future AES and SMPTE standards as they are approved and released, while simultaneously offering backward compatibility with the RAVENNA networking protocol.
Today, 70,000 connected devices worldwide use the Livewire or Livewire+ AES67 protocols, and 100 companies provide compatible equipment. While these numbers are impressive, adoption continues to grow steadily for a number of key reasons.
EASE OF INSTALLATION AND USE
In the coming years, studios’ transition to IP will continue, but the number of available protocols, methods, hardware and audio devices can make the process confusing. Livewire+ AES67 cuts through this noise to accomplish interconnectivity more easily at a lower cost.
Using Livewire+ AES67, uncompressed digital audio, device control messages, program-associated data and routine network traffic is carried over a single Ethernet cable in real time. This reduces the number of cables to deploy, significantly reducing the time required to wire an entire facility. All sources and devices connect using readily available Ethernet cables, which can carry up to 250 audio channels each, depending on the network link capacity. This link aggregation eliminates expensive multi-pair cable for interconnecting studios, resulting in potentially significant savings in labor costs alone.
Configuration can also be just as simple. With Livewire+ AES67, every audio source is given a text name and numeric ID, which are transmitted from devices over the network thanks to a built-in device discovery mechanism. All hardware products have built-in web engines that can be accessed via any common browser or by using an Axia program called iProbe. Users simply enter the names of their desired input sources using just a PC and web browser, with a configuration window enabling any necessary parameter changes for the selected sources.
As a result of these capabilities, installation and configuration, which may have taken weeks in the past, can now be completed in hours thanks to Livewire+ AES67.
AUDIO QUALITY AND RELIABILITY
Unlike Internet audio, which suffers from reduced quality as a result of limited and variable bandwidth, Livewire+ AES67 uses Internet protocols but is intended to deliver uncompressed audio over local area networks (LANs). Livewire+ AES67 is the only fully compliant protocol that also features Unicast SIP modes of operation, meaning it is also suitable for VLAN and WAN applications.
The controlled, high-speed Livewire+ AES67 network provides more than adequate bandwidth for large numbers of channels of high-quality uncompressed audio in real time, eliminating the risk of audio drop-outs from network outages and other issues affecting transmission. Long accepted as a reliable means of transporting virtually any kind of data or signal, IP has become a reliable option for telephone, intercom, teleconferencing and many other applications. According to the Telos Alliance, as of April 2015, more than 5,500 studios around the world had been built using the company’s Axia IP-audio infrastructure employing the Livewire+ AES67 protocol for mission-critical broadcast applications in major metropolitan markets.
A key benefit of Livewire+ AES67 is its ability to enable computer data, phone, audio and control to be transmitted on a single network. Naturally, this type of converged networking environment can generate significant cost-efficiencies throughout a broadcast facility.
Further contributing to the cost-effectiveness of Livewire+ AES67 is that the most widely used and highly respected companies in the radio industry have adopted the technology. This allows broadcasters to select best-of-breed solutions and connect as many audio devices as possible directly to their audio network. In addition to simplicity, this removes the need for extra I/O devices, delivering even lower overall system costs.
Livewire+ AES67 allows wiring to be installed in hours, as opposed to the weeks that are often required. Livewire+ also generates savings from the way it handles audio from PCs, which nearly all broadcast stations use as their primary means of playing and editing audio. With a traditional network, PC-based audio is transmitted through a router input card or console module, which adds significant cost when bringing multiple audio channels into the system.
The latest release of Livewire+ AES67 includes a driver that can handle up to 24 bi-directional stereo or mono audio streams directly through a computer’s network card and now features PTP clock synchronization. This allows the computer to be connected directly to the network using an existing Ethernet connection, eliminating the cost of an additional sound card and the port needed to connect it to a console or router. In many cases, this can save broadcasters many thousands of dollars.
Studios’ transition to IP-based broadcasting has been underway for several years, but has been hindered by the sheer number of protocols, integration methods, legacy hardware and advanced audio devices, which can be confusing at best. The advanced capabilities and other key benefits of Livewire+ AES67, on the other hand, simplifies interconnectivity and upgrades, while offering greater flexibility and cost-effectiveness.
By implementing Livewire+ AES67 solutions, studios can take advantage of best-of-breed technologies to satisfy a wider range of applications within network environments. Best of all, Livewire+ AES67 is designed to work with future standards as they are released. This ensures that the Livewire+ AES67 will never be obsolete and continue to deliver quality, reliability, flexibility, cost savings and many more benefits well into the future.
John Schur is the president of Telos Alliance TV Solutions Group.
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