How AES67 Solves the AoIP Compatibility Problem

Modern broadcast facilities rarely rely on a single vendor. A typical news production center might use a Lawo routing matrix, Wheatstone mixing consoles, Sennheiser microphone receivers, RTS intercom panels, and Tieline codecs—each with its own native audio networking protocol. Without a common interoperability standard, integrating these systems would require purpose-built gateways, custom software wrappers, or costly format conversion at every interface point. AES67 eliminates this friction by defining a universal transport layer that any compliant device can speak.

Developed by the Audio Engineering Society and formally published as AES67-2015 (reaffirmed in 2018), the standard specifies the protocols and parameters needed for high-performance audio-over-IP streaming between devices regardless of manufacturer. It is not a complete networking framework but rather a compatibility layer—a set of constraints that existing AoIP technologies can adopt to communicate with one another. This approach means AES67 does not replace proprietary protocols like Dante, RAVENNA, or Livewire; instead, it provides a common denominator that these systems can use to bridge their differences.

For broadcasters transitioning from legacy SDI-based infrastructures to IP-native workflows, AES67 serves as the interoperability backbone. It allows a console from one vendor to send audio to a router from another, or a microphone receiver to stream directly to a codec, all over standard Ethernet without intermediate format conversion. This article explores the technical underpinnings of AES67, its real-world applications in multi-vendor environments, and its strategic role in the broader industry shift toward all-IP production.

Core Protocols and Layers

AES67 leverages established transport protocols to ensure low latency, high reliability, and synchronization. The standard mandates the use of RTP (Real-time Transport Protocol) for carrying audio data, with PCM (Pulse-Code Modulation) encoded audio at sampling rates up to 96 kHz. For stream discovery and session management, AES67 relies on SAP (Session Announcement Protocol) to announce streams on the network. Timing and synchronization are achieved through IEEE 1588-2008 Precision Time Protocol (PTP), which provides sub-microsecond accuracy across networked devices. This combination of protocols creates a foundation that is both standards-based and widely supported by network infrastructure.

How AES67 Works: The Technical Foundation

At its core, AES67 operates by defining a set of mandatory parameters that all compliant devices must support. These include sample rates (48 kHz and 96 kHz), bit depth (16, 24, or 32 bits), and packet times (1 ms, 125 μs). By locking these parameters, AES67 ensures that any two compliant devices can exchange audio with deterministic latency and synchronized timing. The standard also specifies the use of PTP profiles—specifically the SMPTE ST 2059-2 profile—to align audio streams across a shared reference clock.

In practice, a broadcast facility may have a mixing console from one vendor that natively supports its own AoIP format but also includes an AES67 mode. When the console is set to output AES67-compatible streams, those streams can be received by an AES67-compliant microphone preamplifier, a codec, or a studio router from a completely different manufacturer. The key enabler is the PTP clock, which synchronizes all endpoints to the same timebase. Without this common clock, jitter and drift would render the audio streams unusable in a live production context.

Packet Timing and Latency Considerations

AES67 defines three packet time options: 1 ms, 125 µs, and 250 µs. Shorter packet times reduce latency but increase network overhead because more packets must be processed per second. A 1 ms packet time means 1000 packets per second per stream, while 125 µs means 8000 packets per second. Broadcasters must balance the latency requirements of their application—live production typically demands sub-5 ms end-to-end latency—against the processing capacity of their network switches and endpoints. For most broadcast applications, 1 ms packet time offers a good compromise between low latency and manageable network load.

The standard also specifies a maximum transmission unit (MTU) of 1500 bytes for IPv4 networks, ensuring compatibility with standard Ethernet infrastructure. Each audio packet contains a sequence number and timestamp, allowing the receiver to reconstruct the original audio waveform even if packets arrive out of order due to network jitter. The RTP payload format follows the L24 standard for linear PCM audio, with channels packed sequentially within each packet.

Interoperability in Practice: Real-World Multi-Vendor Workflows

The true value of AES67 becomes apparent in complex broadcast environments where no single vendor can supply every component. Consider a large news production center that uses a Calrec mixing console (which uses its own AoIP protocol but also supports AES67), Sennheiser wireless microphones with an AES67-compatible receiver, and a Lawo audio router. Without AES67, integrating these systems would require expensive gateways or custom software to translate between proprietary formats. With AES67, each device can be configured to send and receive audio using the standard's RTP and PTP settings, allowing the broadcast engineer to route audio sources directly from microphones to the console without intermediate conversion.

Another common use case is remote contribution. A field reporter may use a portable AES67-compatible codec (such as those from AETA or Tieline) to send high-quality audio back to the studio. In the studio, an AES67-compliant matrix can receive the stream and route it to the on-air mixing console, all over a shared IP network. This eliminates the need for dedicated analog or AES3 connections and provides the flexibility to reconfigure signal paths via software.

Reducing System Integration Costs

By adopting AES67 as a common interoperability language, broadcasters can avoid the "vendor lock-in" that often accompanies proprietary ecosystems. System integrators can design infrastructure that is agnostic to the final equipment choices, allowing broadcasters to source components based on features and price rather than compatibility concerns. This competition drives innovation and often reduces overall system costs.

For example, a station upgrading its audio infrastructure might choose an AES67-compliant router from a specialized manufacturer, while selecting microphones and intercom panels from other vendors. The only requirement is that each device supports AES67, which many professional broadcast audio products now do. This modular approach makes it easier to scale the system over time, adding new hardware without overhauling the entire network.

Case Study: A Regional Broadcaster's Transition to AES67

Consider a regional broadcaster that operates two studios and a master control room. Historically, the facility used analog audio routing with a central patch bay. After transitioning to an all-AES67 network, the broadcaster was able to install a mixing console from vendor A, a routing matrix from vendor B, and codecs from vendor C—all interoperable. The IT department managed the network switch configuration, while the engineering team used a unified management tool to assign audio streams. The result was a 30% reduction in cabling and a 50% increase in routing flexibility.

During the transition, the broadcaster worked with a system integrator to segment the audio network using VLANs and to configure PTP boundary clocks on each switch. The engineering team discovered that some older devices required firmware updates to achieve full AES67 compliance. After testing with an AES67 analysis tool, they confirmed that all endpoints met the standard's timing requirements. The project was completed within budget and on schedule, demonstrating that migration to AES67-based audio is both technically and economically feasible for smaller operations.

AES67 vs. Other AoIP Standards

To fully appreciate AES67's role, it helps to compare it with other major AoIP protocols:

  • Dante (Audinate): The most widely deployed AoIP system in professional audio. Dante devices natively communicate using a proprietary protocol but can optionally support AES67 for interoperability. Dante offers management features like automatic device discovery and advanced routing via Dante Controller.
  • RAVENNA (ALC Network): An open standard used primarily in broadcast and recording. RAVENNA is built on many of the same underlying protocols as AES67 and fully supports it. Many RAVENNA devices are AES67-compliant out of the box.
  • Livewire+ (Telos Alliance): Popular in radio broadcast, Livewire+ is a proprietary protocol that has integrated AES67 support in newer devices, allowing cross-vendor connectivity.
  • SMPTE ST 2110 Suite: A set of standards for professional media over managed IP networks, covering video, audio, and ancillary data. ST 2110-30 for audio is essentially AES67 wrapped in the SMPTE framework. Therefore, devices supporting ST 2110 are AES67-compatible by necessity.

AES67 sits at the intersection of these systems. It does not attempt to replicate their management and control features but instead provides a guaranteed minimum capability set. As a result, a Dante device with AES67 enabled can send audio to a RAVENNA device without needing a Dante-to-RAVENNA gateway—both speak AES67.

The NMOS Control Layer

While AES67 excels at transport, it leaves discovery and connection management to individual vendors. This means a device from one manufacturer may not be discoverable by the management software of another, even though both support AES67. To address this gap, the AES67 standard is increasingly paired with NMOS (Network Media Open Specifications), developed by the Advanced Media Workflow Association (AMWA). NMOS provides APIs for device discovery (IS-04), connection management (IS-05), and timing (IS-06). The combination of AES67 and NMOS—often called "AES67+NMOS"—is emerging as a more complete solution for broadcast IP control, enabling the kind of plug-and-play interoperability that engineers expect from legacy SDI infrastructure.

Benefits for Broadcasters and System Integrators

The adoption of AES67 brings tangible advantages to broadcast operations, from flexibility to cost savings:

  • Vendor Independence: Equipment from different manufacturers can be mixed and matched, future-proofing investments.
  • Operational Agility: Reconfiguring audio routing becomes a software change rather than a physical patch, enabling rapid setup for breaking news or multi-format production.
  • Reduced Complexity: Fewer conversion boxes and proprietary interfaces mean fewer points of failure and easier troubleshooting.
  • Scalability: IP networks can be expanded incrementally; adding an AES67-compliant device is usually a matter of connecting it to the switch and configuring its stream parameters.
  • Interoperability with Video: Through SMPTE ST 2110, AES67 audio can be seamlessly integrated with IP video streams, simplifying hybrid production environments.
  • Future-Proof Infrastructure: AES67-compatible equipment can be redeployed as the facility evolves, protecting capital investments.

Testing and Certification Considerations

Not all devices labeled "AES67-compatible" are created equal. The standard defines mandatory parameters, but some manufacturers implement optional features inconsistently. For critical broadcast applications, engineers should verify compliance using test tools such as the AES67 Analyzer from the EBU or the AES67 Conformance Tool from the Audio Engineering Society. These tools validate timing accuracy, packet formatting, and PTP synchronization. In addition, many vendors participate in interoperability testing events, such as the AES67 Plugfest, where devices from different manufacturers are tested together in a controlled environment. Specifying devices that have passed formal conformance testing reduces integration risk.

Challenges and Considerations

While AES67 is a powerful tool, it is not without challenges. Because AES67 focuses only on the transport layer, management and control functions are left to individual vendors. This means that while devices can exchange audio, they often require separate configuration interfaces. A device from one manufacturer may not be discoverable by the management software of another. To address this, the AES67 standard is sometimes paired with NMOS (Network Media Open Specifications), which provides discovery and registration APIs. The combination of AES67 and NMOS (called "AES67+NMOS") is emerging as a more complete solution for broadcast IP control.

Another consideration is network performance. AES67 demands low latency and high precision timing. Switches must support IEEE 1588 boundary clocks or transparent clocks to maintain PTP accuracy across large networks. Unmanaged or insufficiently configured switches can introduce jitter that degrades audio quality or causes dropouts. Broadcast engineers must work closely with network specialists to design a properly segmented IP infrastructure with Quality of Service (QoS) policies that prioritize multicast audio traffic.

Additionally, AES67 does not specify a compression codec—all streams are uncompressed. While this ensures high audio quality, it consumes significant bandwidth. A single 48 kHz, 24-bit stereo stream uses about 2.3 Mbps. In a large facility with hundreds of streams, careful capacity planning is essential. Using multicast rather than unicast reduces network load because multiple receivers can subscribe to the same stream without duplicating traffic. However, multicast requires IGMP snooping on network switches to prevent unnecessary flooding.

Migration Strategies from Legacy Systems

Broadcasters transitioning from SDI-based audio to AES67-based AoIP have several options. The simplest approach is a phased migration, where AES67-compatible devices are added alongside existing analog and AES3 infrastructure. Gateway devices such as the AES67-to-AES3 converters allow legacy equipment to participate in the IP network. Over time, as older devices are retired, the facility becomes fully IP-native. Another approach is a full cutover, where the entire audio infrastructure is replaced at once. This is riskier but can be completed during a planned downtime window, such as a station outage for transmitter maintenance. Most broadcasters choose the phased approach because it spreads capital expenditure and allows engineers to gain experience with IP networking gradually.

The Future of AES67 in IP-Based Broadcast

AES67 is not static. The standard is maintained by the Audio Engineering Society and continues to evolve alongside industry needs. The recent publication of AES67-2018 clarified several technical points and expanded support for higher sample rates. Looking ahead, AES67 is expected to remain the backbone for audio interoperability in the growing ecosystem of ST 2110 deployments. As more manufacturers adopt NMOS for control, the seamless "plug-and-play" vision for AoIP becomes closer to reality.

Furthermore, AES67 is increasingly used beyond traditional broadcast—in live sound, corporate AV, and even automotive telematics. Its open nature makes it attractive for any application requiring deterministic audio transport over standard Ethernet. The standard has also been adopted by the European Broadcasting Union (EBU) as Tech 3326, which provides additional guidelines for broadcast-specific implementations. This institutional backing ensures that AES67 will remain relevant as the industry moves toward all-IP production.

For broadcasters planning their IP transition, investing in AES67-compatible equipment now ensures that future expansions and upgrades will integrate smoothly, regardless of vendor choices. The standard's longevity is underpinned by its adoption by major industry bodies, including the EBU (Tech 3326) and SMPTE (ST 2110-30).

Conclusion

AES67 is more than a technical specification; it is a catalyst for change in the broadcast industry. By providing a universal bridge between proprietary AoIP systems, it enables the multi-vendor interoperability that modern broadcasters demand. While challenges remain—particularly in management and network design—the benefits of flexibility, cost savings, and future-proofing are undeniable. As the industry continues its migration toward all-IP infrastructures, AES67 will stand as a cornerstone of open, interoperable, and high-performance audio networking.

For further reading, consult the official AES67 standard document, the EBU Tech 3326 recommendation, and the SMPTE ST 2110 standards suite for professional media over IP networks. Additional resources include the AMWA NMOS specifications for discovery and connection management, and the Audinate Dante documentation for details on enabling AES67 in Dante deployments.