The shift toward remote production and distributed audio workflows has accelerated dramatically in recent years, driven by the need for flexibility, cost efficiency, and global collaboration. At the heart of this transformation lies AES67, an open standard that bridges the gap between proprietary audio-over-IP (AoIP) systems. By enabling interoperability across diverse equipment and networks, AES67 has become a foundational technology for broadcasters, recording studios, and live event producers operating in increasingly decentralized environments. This article explores the mechanics of AES67, its profound impact on remote production, the practical realities it introduces for distributed teams, and what the future holds as the standard continues to evolve alongside network advancements.

What Is AES67?

AES67 is a standard developed by the Audio Engineering Society that specifies how different AoIP systems can exchange audio streams over standard IP networks. Unlike proprietary protocols (such as Dante, Ravenna, or Livewire), AES67 defines a common set of transport, synchronization, and media format rules. This means that a microphone preamplifier using one protocol can send audio directly to a mixing console that speaks a different protocol, as long as both implement AES67. The standard is not a complete protocol itself but rather a compatibility layer; existing AoIP systems gain AES67 support to open interoperability doors.

Core Technical Components

AES67 defines four key areas:

  • Synchronization – Uses Precision Time Protocol (PTPv2, IEEE 1588-2008) to keep all devices on the network sample-accurately aligned. This ensures that audio streams from different sources arrive with minimal jitter and no drift, critical for multi-track recording and live mixing.
  • Media Transport – Relies on Real-time Transport Protocol (RTP) over UDP for streaming audio. The default payload format is Linear PCM, with sample rates up to 96 kHz and bit depths up to 24 bits. Support for compressed formats is optional.
  • Discovery and Connection Management – The standard provides a framework using Session Description Protocol (SDP) to describe audio streams. Devices advertise their capabilities and can be connected via external control systems, but AES67 itself does not mandate a specific discovery protocol, allowing integration with existing systems like Dante Controller or Ember+.
  • Quality of Service (QoS) – AES67 specifies packet times (1 ms or 125 µs) and requires network infrastructure capable of handling isochronous traffic. Priority tagging (DiffServ) is recommended to ensure low latency and no packet loss.

By standardizing these layers, AES67 allows a single network to carry audio from multiple manufacturers without dedicated interfaces. This openness reduces vendor lock-in and simplifies system design for remote production hubs.

How AES67 Transforms Remote Production

Remote production involves capturing, processing, and broadcasting audio from locations that are not physically co-located with the main control room. Traditional methods required dedicated analog or digital lines (such as ISDN or satellite circuits), which were expensive and inflexible. AES67, running on standard Ethernet and IP networks, enables broadcasters to connect studios, OB vans, field reporters, and production centers using commodity network gear.

High-Quality Low-Latency Audio

Latency is the enemy of live production. Singers performing remote duets, talk show hosts taking calls, or sports commentators reacting to live action all require sub‑10 ms round-trip delay. AES67 achieves this through precise PTP synchronization and efficient RTP transport. A typical AES67 stream adds only 1–2 ms of network delay per hop, making it suitable for even the most latency-sensitive applications like live mixing or foldback monitoring. For comparison, older IP audio codecs often introduced 20–40 ms of delay, which disrupted natural interaction.

Cost Reduction and Scalability

By using existing IT infrastructure, AES67 eliminates the need for expensive proprietary routers, break-out boxes, and dedicated audio snakes. Remote production rigs can be assembled with off-the-shelf network switches and standard Cat6 cabling. Scaling a remote production—adding more microphones, mix-minus feeds, or intercom channels—requires only additional network ports and configuration changes, not new hardware. This agility is especially valuable for broadcasters covering unpredictable events like breaking news or multi‑venue festivals.

Simplified Connectivity Across Diverse Gear

In a typical remote truck, you might find a mixing console from one manufacturer, a stagebox from another, and a wireless mic system from a third. Without AES67, each device likely speaks its own AoIP protocol, requiring expensive interfacing gateways or multiple network segments. AES67 allows all gear to share a single audio network, provided the manufacturers have implemented the standard. This interoperability reduces cabling complexity, minimizes signal conversion points (which can degrade quality), and accelerates setup times for remote crews.

Enabling Distributed Audio Workflows

Distributed audio workflows involve multiple contributors working on the same project from different locations. A sound designer in Los Angeles, a foley artist in Berlin, a re-recording mixer in Sydney—all need to access the same high-resolution audio streams with minimal latency. AES67 makes this practical by standardizing the transport layer across wide-area networks (WANs).

Seamless Multi‑Site Collaboration

When AES67 is combined with a managed WAN or VPN, musicians and engineers can send channels of uncompressed audio between sites as if they were in the same room. A classical music ensemble can record each section in separate cities, then merge the tracks in real time for a virtual ensemble performance. Film and TV post‑production teams can stream stems to remote mixing suites without sacrificing sample accuracy. The standard’s PTP synchronization extends across network boundaries via boundary clocks, ensuring that all locations maintain sample alignment to within microseconds.

Scalability for Growing Productions

Distributed workflows often need to expand rapidly. A podcast network adding new hosts in different cities, or a live event company acquiring a second broadcast unit, can integrate additional AES67 devices into the existing network without re-engineering the whole system. The standard supports multiple redundant streams and can handle hundreds of audio channels on a single 1 Gbps link, making it feasible to build large-scale, multi‑site audio infrastructures with predictable performance.

Synchronization Accuracy for Professional Output

One of the biggest challenges in distributed audio is keeping all signals perfectly timed. Even a few samples of drift between two remote sources can cause phasing issues, lip-sync errors, or misalignment when tracks are later edited. AES67 uses PTPv2 with hardware timestamping to achieve sub‑microsecond synchronization. This is far more precise than consumer‑grade NTP, and it meets the requirements of broadcast‑grade production. Facilities no longer need a dedicated house sync distribution; the network itself provides the clock.

Practical Considerations for Implementation

While AES67 offers powerful capabilities, successful deployment demands careful attention to network design and configuration.

Network QoS and Bandwidth Planning

AES67 streams are isochronous and sensitive to packet loss. A single dropped packet can cause an audible click or dropout. Engineers must configure their switches with strict QoS policies, prioritizing AES67 traffic (typically using DiffServ EF or AF41). The network should be dedicated or have a clearly separated VLAN for audio to avoid congestion from data traffic. Bandwidth requirements are straightforward: a single 48 kHz, 24‑bit stereo stream uses about 6 Mbps; 64 channels at 96 kHz would need roughly 200 Mbps. Gigabit Ethernet is sufficient for most applications, but larger facilities may require 10 GbE backbones.

Device Configuration and Compatibility

Not all devices that advertise “AES67” support every optional feature. For example, some may not implement PTP boundary clock mode, limiting their use in complex multi‑subnet networks. Others may require specific firmware versions. Before deployment, it is wise to test interoperability between your specific mix of gear. Many manufacturers provide AES67 compatibility guides or lists of tested partner devices. Network engineers should also understand the device’s SDP configuration—some require manual IP assignment, while others support automatic discovery via SAP or mDNS.

When extending AES67 over wide‑area networks, additional latency from routing, switching, and propagation delay must be accounted for. A typical cross‑continent link (e.g., New York to London) adds roughly 50–80 ms round‑trip. While too high for live foldback, this is acceptable for recording or broadcast contribution where the audience sees a delay anyway. For real‑time collaboration, producers should keep WAN hops to a minimum and use dedicated circuits or SD-WAN with deterministic routing.

Real‑World Applications: Broadcast, Music, and Live Events

AES67 is not a theoretical standard; it is deployed today in thousands of facilities worldwide.

Broadcast and News Production

Major broadcasters like the BBC and NPR use AES67 to connect remote studios, portable field units, and central production centers. During the pandemic, many radio networks rapidly set up home studios for presenters using AES67-compatible desktop interfaces. This allowed hosts to send high‑quality audio to the studio from their homes, while still receiving IFB mixes and talkback via the same network. The standard’s low latency kept live conversations natural, and its interoperability meant that presenters could use a variety of consumer‑friendly devices.

Music Recording and Post‑Production

In the music industry, AES67 enables virtual recording sessions where tracking, overdubbing, and mixing happen across cities. Some commercial studios now offer “remote recording pods” that connect to their main control room via AES67 over the internet. The standard has also been adopted by certain digital audio workstations (DAWs) and audio interfaces, allowing a producer in one location to stream multiple channels of audio directly into the timeline of an engineer in another location, with sample‑accurate alignment.

Live Event Streaming

Large festivals and live sports events increasingly use AES67 to connect numerous microphones, IEM systems, and broadcast feeds across sprawling venues. Rather than running hundreds of analog cables, the production team builds a single network with multiple switch drops. AES67 streams carry stage audio to FOH, monitors, broadcast trucks, and recording rigs simultaneously. If the event is also being streamed online, the same network can deliver clean feeds to remote encoding stations, simplifying signal distribution.

Future Prospects: AES67, ST 2110, and Beyond

AES67 is already being integrated into broader standards like SMPTE ST 2110, which extends the same principles to video and metadata. Many broadcasters transitioning to IP infrastructure (SMPTE ST 2110, NMOS, etc.) use AES67 as the audio component. This convergence means that AES67 will remain relevant for years, even as networks move to higher data rates and more advanced synchronization schemes.

5G and Edge Computing

The rollout of 5G networks promises low‑latency, high‑bandwidth wireless connections that are ideal for AES67. Field reporters could transmit several channels of uncompressed audio from a 5G‑enabled backpack to the studio without the need for dedicated satellite links. Edge computing nodes placed near transmission towers could process PTP sync locally, further reducing jitter. This could enable truly mobile remote production, where the control room is anywhere.

Cloud‑Based Production

Cloud audio production is gaining traction, with solutions like AWS Elemental MediaConnect and Azure Media Services supporting AES67. Producers are beginning to route audio from on‑premise consoles to cloud servers for post‑production or AI processing. The standard’s strict synchronization model is being adapted for virtualized environments, though challenges around clock distribution in the cloud remain an area of active development.

The Role of AES67 in Education

For students and educators researching audio technology, AES67 represents a shift toward open, network‑based audio. Many universities now include AoIP and AES67 in their curriculum, teaching future sound engineers and broadcasters how to design, troubleshoot, and optimize these networks. Understanding AES67 is becoming as fundamental as learning microphone placement or mixing techniques.

Conclusion

AES67 has fundamentally changed the economics and possibilities of remote production and distributed audio. By providing a common language for disparate AoIP systems, it has lowered barriers to entry for high‑quality audio networking, enabling workflows that were previously impractical or prohibitively expensive. As the need for remote collaboration continues to grow, AES67 will likely remain the backbone that connects talent, equipment, and audiences across the globe. Its continued integration with video standards and emerging network technologies ensures that audio professionals—whether in broadcasting, music, or live events—will rely on it for years to come.

For further reading, consult the AES Standards website, the SMPTE ST 2110 documentation, or industry resources such as Audio Media International and Ravenscroft Audio coverage of AoIP deployments.