The AES67 standard is a critical development in the broadcasting industry, enabling interoperable audio-over-IP networks. It allows different manufacturers' equipment to communicate seamlessly, fostering a more flexible and efficient broadcasting environment. As broadcast facilities transition from traditional point-to-point analog and digital audio connections to networked IP-based infrastructure, AES67 serves as a foundational technology that ensures diverse devices can work together on a common network. This article provides an in-depth exploration of AES67, its technical underpinnings, implementation considerations, and its role in modern broadcasting workflows.

What is AES67?

AES67 is an open standard developed by the Audio Engineering Society (AES). It specifies a common framework for high-performance audio-over-IP and audio-over-Ethernet networks. The goal is to ensure interoperability between devices from various manufacturers, reducing compatibility issues and streamlining production workflows. Officially designated as AES67-2018 (and recently updated), the standard defines a set of protocols and parameters for transporting digital audio over IP networks with low latency and high synchronization accuracy. It was first published in 2013 and has since been widely adopted by broadcast equipment manufacturers, audio networking vendors, and professional audio system integrators.

The standard addresses the problem of fragmented audio networking protocols. Prior to AES67, each manufacturer often developed its own proprietary system—Dante from Audinate, Ravenna from ALC Network, Livewire from Wheatstone, and others. While these systems are powerful within their ecosystems, they do not naturally interoperate. AES67 provides a common denominator: a set of mandatory requirements that any compliant device must support, enabling a device from one vendor to send or receive audio streams with a device from another vendor.

Key Features of AES67

AES67 is built around several core features that together allow reliable, low-latency, and synchronized audio transport over standard Ethernet networks.

Interoperability

The primary feature of AES67 is interoperability. It defines a common stream format (linear PCM audio at standard sampling rates like 48 kHz) and uses established protocols such as Real-time Transport Protocol (RTP) for encapsulating audio data. By specifying mandatory codec parameters (e.g., L24 uncompressed audio, 24-bit depth) and session description format (SDP), AES67 ensures that any compliant device can decode and play audio from another. This cross-vendor compatibility is essential for broadcasters who need to integrate systems from different suppliers without requiring custom gateways or converters.

Low Latency

AES67 supports real-time audio transmission suitable for live broadcasting. Typical network latency is measured in sub-millisecond to a few milliseconds, depending on network design and buffering. The standard specifies a maximum end-to-end latency of 1 millisecond for point-to-point connections and up to 10 milliseconds for complex switched networks. Low latency is achieved through careful timing and small packet sizes (e.g., 1 ms audio payload per packet). Broadcast engineers must design their networks with sufficient bandwidth and low jitter to maintain these latency figures.

Synchronization

AES67 uses Precision Time Protocol (PTP, IEEE 1588-2008, also known as PTPv2) to synchronize audio streams precisely. PTP provides a clock distribution mechanism that aligns the sample clocks of all devices on the network to a common grandmaster clock. This synchronization is required to ensure that audio samples from different sources are aligned when mixed or processed. Without precise timing, audio would drift, leading to glitches, pops, or loss of phase coherence. AES67 mandates support for PTP profiles such as the AES67 profile (defined in annex A of the standard) and also the SMPTE ST 2059-1 profile for use in broadcast facilities that also carry video over IP.

Standardized Protocols

AES67 implements standard protocols for streaming and discovery. The audio data is transported using RTP (Real-time Transport Protocol) over UDP/IP. Session description is done via SDP (Session Description Protocol), which contains all parameters needed by a receiver to lock onto and decode a stream: IP addresses, port numbers, payload type, sampling rate, number of channels, and timing information. Discovery of AES67 streams is not part of the standard itself; instead, manufacturers can use existing discovery protocols such as SAP (Session Announcement Protocol), mDNS, or proprietary methods. However, many AES67 implementations also support the IEEE 1722 Media Clock Recovery mechanism for clock synchronization when PTP is not available.

How AES67 Works in Broadcasting

In a typical broadcast setup, AES67 enables various audio devices—such as microphones, mixers, and recording systems—to connect over an Ethernet network. The standard manages the timing, synchronization, and data transfer, ensuring high-quality audio with minimal delay. This setup allows broadcasters to create flexible, scalable, and reliable audio networks that can be reconfigured easily.

Network Architecture

AES67 networks are built on standard Ethernet infrastructure, but they require careful planning to meet timing and bandwidth demands. Broadcast engineers typically use managed switches that support IGMP snooping (for multicast) and Quality of Service (QoS) to prioritize audio traffic. VLANs can segregate audio streams from other data traffic to reduce contention. The network must be designed with sufficient capacity (usually 1 Gbps or higher) and low latency (sub-millisecond switch latency). Redundancy can be achieved through redundant switches and link aggregation, though AES67 itself does not define a specific redundancy mechanism; broadcasters often use SMPTE ST 2022-7 for seamless redundancy.

Audio Streams

AES67 supports both unicast (point-to-point) and multicast (one-to-many) transmission. Unicast is simpler but uses separate network flows for each receiver; multicast is more efficient for distributing a single audio stream to multiple destinations (e.g., sending program audio to many console inputs). The sender selects a multicast IP address and uses IGMP to manage group membership. SDP parameters inform receivers which multicast address to join. Bandwidth per stream depends on sample rate, bit depth, and number of channels; a typical 48 kHz, 24-bit, 8-channel stream consumes about 9.2 Mbps.

Advantages of Using AES67

  • Flexibility: Easily integrate equipment from different manufacturers without proprietary gateways.
  • Cost-Effective: Reduces the need for proprietary hardware and dedicated wiring; leverages existing IT infrastructure.
  • Future-Proof: Supports ongoing technological advancements in IP audio; alignment with SMPTE ST 2110 for video+audio transport.
  • Enhanced Workflow: Simplifies setup and management of complex audio networks via software configuration.
  • Scalability: Add new devices without massive rewiring; use IP routing to send audio anywhere on the network.

Challenges and Considerations

Despite its advantages, implementing AES67 requires careful planning. Network security, proper synchronization, and bandwidth management are vital for optimal performance. Additionally, not all equipment may fully support AES67, necessitating compatibility checks before deployment.

A major challenge is network design for low jitter. Ethernet switches introduce variable latency, and IP routing can cause packet delay variation. To counteract this, AES67 devices employ jitter buffers, which add latency but smooth out timing irregularities. Engineers must balance buffer size against acceptable end-to-end delay (e.g., for live on-air use, total latency should be under 5 ms). Another consideration is discovery: while AES67 standardizes stream transport, it does not prescribe a unified discovery method. A receiver cannot automatically find an AES67 stream unless both sender and receiver use the same discovery protocol (e.g., both support SAP or both use a vendor's proprietary discovery). This can complicate plug-and-play setups.

Security is also an issue. AES67 streams are typically not encrypted; any device on the network can join a multicast group and receive audio. Broadcasters should use VLANs, firewalls, and potentially IPsec or other encryption to protect sensitive content. Additionally, PTP synchronization relies on a reliable network; loss of PTP packets can cause clock drift and audio break-ups. Network redundancy and careful PTP boundary clock configuration are recommended.

Comparison with Other AoIP Standards

AES67 is often compared to Dante from Audinate, Ravenna from ALC Network, and Livewire+ from Wheatstone. While all three are capable IP audio networks, AES67 is the only one that is an open standard (not proprietary) and is designed for multi-vendor interoperability. Dante uses proprietary discovery, clocking, and stream management, but many Dante devices can be configured to output or receive AES67 streams (via Dante Controller's "AES67" mode). Ravenna is natively compliant with AES67 and also supports the higher-channels-per-packet efficiency of RAVENNA's own streaming. Livewire+ is used extensively in broadcast radio and supports AES67 as an optional transport mode.

For broadcasters, the choice often depends on the existing ecosystem. If a facility is already all-Dante, AES67 may be used to connect to other AES67-only devices. New facilities may start with Ravenna or AES67-natively to avoid vendor lock-in. The SMPTE ST 2110 suite for professional media over IP relies heavily on AES67 for its audio component (ST 2110-30), making AES67 essential for any facility moving toward full IP video/audio production.

Use Cases in Broadcasting

AES67 is deployed in a variety of broadcast applications:

  • Live Radio and TV Studios: Connecting microphones, mixing consoles, codecs, and playout systems over a common network.
  • Inter-Facility Links: Transporting program audio between production studios and transmission sites via IP WAN links, using AES67 over RTP.
  • Remote Production: Returning audio from remote locations using IP codecs that support AES67, ensuring compatibility with the home studio's infrastructure.
  • Multi-Room Audio Distribution: Sending clean feed, mix-minus, and monitoring signals across multiple studios in a building without dedicated audio cabling.
  • Integration with Video-over-IP: In SMPTE ST 2110 facilities, AES67 audio streams are carried alongside video and ancillary data, synchronized via common PTP.

Future of AES67

The AES67 standard continues to evolve. Recent revisions have clarified media clock recovery and PTP profile requirements. The standard is also being adopted beyond broadcast into professional sound reinforcement, installed sound, and live event production. As the industry moves toward fully IP-based production, AES67's role as a common transport for audio is secure. The AES67-2018 standard is now widely referenced by broadcast equipment manufacturers, and certification programs (such as the AES67 Interoperability Certification) help ensure that products work together out of the box.

Looking ahead, AES67 will likely merge or converge with newer standards like SMPTE ST 2110-30 (which is AES67-compliant). The Audio Engineering Society and SMPTE have collaborated to align their profiles, making it easier to build workflows that carry both video and audio over IP. For broadcasters, investing in AES67-compatible equipment is a long-term strategy that supports both today's multi-vendor needs and tomorrow's full-IP production environments.

Conclusion

The AES67 standard is transforming the broadcasting industry by enabling interoperable, flexible, and high-quality audio-over-IP networks. As technology advances, adopting AES67 will become increasingly essential for broadcasters aiming to stay competitive and innovative in a digital landscape. By providing a common language for audio transport, synchronization, and latency control, AES67 simplifies network design, reduces costs, and opens the door to seamless integration of best-of-breed equipment from multiple manufacturers. Broadcast engineers and facility managers should consider AES67 compatibility a key criterion when selecting audio networking products today.

For further reading, see the official AES Standards page for AES67 documentation, the RAVENNA website for AES67-compliant implementations, and this IEEE overview of PTP for deeper synchronization understanding. Also refer to EBU Tech 3326 for best practices in IP audio networking.