Today's broadcasters are seeing a steady increase in the number of audio and video channels in use. Controlling and routing all of these signals using legacy 
non-network transports can be a daunting task, involving expensive specialized routers and complex workflows that vary from one manufacturer to another. Adding and incorporating new equipment can mean upgrading and replacing many other pieces of gear to maintain compatibility and provide bandwidth. By way of contrast, an IP-based solution can handle many hundreds or thousands of channels of audio, connecting dozens of devices using inexpensive Cat5E cabling and a few inexpensive gigabit network switches. There are no specialized routers needed to provide conversion and distribution; all changes are made quickly and easily in software running on ordinary computers, with no disruption of production activites. {mosimage}

Gigabit and faster network speeds have made IP networking an indispensable medium. In broadcast, it is simply the best way to transmit bit-perfect audio between as many devices as needed, with low latency and tight synchronization. Realistic, functional interoperability is required to allow facilities to use products they prefer even if they employ different 
audio-over-IP technology.

AES67 is a recent standard that seeks to accomplish this task. The AES67 standard is a networked audio interoperability specification developed by the Audio Engineering Society. It describes techniques for exchanging digital audio on a TCP/IP network using RTP, or real-time transport protocol. Additionally, AES67 specifies particular implementation constraints to facilitate interoperability between implementations.

It is important to note that AES67 is not a complete audio networking solution and does not include all the components required for that role. Technologies provide the layers of discovery, routing, diagnostics, 
auto-configuration, software, and support needed to form a workable audio networking solution for both users and manufacturers. In contrast, a system that uses AES67 to connect multiple network solutions will still require separate management tools for each solution in order to control devices, making the setup much more complex and error-prone than a single-solution system.

AES67 promises basic interconnectivity at its core, and has fairly modest and achievable goals. Primarily focused on the audio networking transport element, AES67 does not specifically address system control, signal routing, or channel labeling. For audio quality, it requires 48 kHz, 24-bit stream with one-millisecond latency as a lowest common denominator. AES67 allows for, but does not require, support for higher sample rates and different bit depths, which means that supported audio formats may vary between devices in the ecosystem. Since AES67 is essentially a set of network standards around how audio channels move across an IP network, it represents a pragmatic evolution in audio networking. Unlike previous specifications (e.g., AVB), AES67 offers a standards-based way to deliver multichannel audio between devices across a network without requiring specialized network equipment. This is significant as it is the first specification to achieve this goal. The AES67 transport operates with common off-the-shelf switches in Layer 3 architecture. Therefore, AES67 deployments will not face the adoption delays that have challenged AVB rollouts.

In creating the AES67 interoperability standard, the AES organization utilized proven existing standards, which mitigate the risk of moving to networked audio.

AES67 does not replace complete audio networking solutions, but enhances them by providing a standards-driven approach for useful, low-level interconnection with others. Leading audio-networking solutions will continue to provide the features necessary for reliable, complete systems that are easy to use and understand, including 
matrix-style signal-routing software, virtual soundcards, network health and clock status monitoring, real-world naming for devices and channels.

In contrast, AES67 only specifies the baseline connectivity of audio streams, and more closely resembles an audio-over-IP version of MADI or AES3. It focuses on how audio channels move through the network between points without defining how routing or switch-defined control may occur.

It is important that companies in this area continue to be involved in the development and tracking of new standards and protocols that further the possibilities of audio networking for TV and radio broadcasting. The industry will continue to develop new standards that require implementation in a sensible way, and it is important to have robust networking solutions that can evolve to incorporate the latest standards.