The adoption of AES67, an audio-over-IP interoperability standard, is fundamentally reshaping broadcast and studio environments. Originally developed by the Audio Engineering Society, AES67 provides a framework for high-performance audio streaming over IP networks, enabling devices from different manufacturers to communicate seamlessly. As the industry moves toward fully networked infrastructures, the standard's role becomes increasingly central. While early adoption focused on large-scale broadcasters, current momentum suggests a rapid expansion into smaller studios, production trucks, and even live music venues. This article explores the key trends that will drive AES67 adoption over the next five years, from deeper integration with video-over-IP standards to the emergence of simplified certification programs that lower barriers to entry.

Current Landscape of AES67 in Broadcast and Studio Environments

To understand future trends, it is essential to assess where AES67 stands today. The standard, first published in 2013, specifies a common set of transport, timing, and media format parameters so that audio-over-IP devices can interoperate at the network level. Early implementations were primarily found in high-end broadcast facilities and by early adopters within the pro-audio community. Today, AES67 is widely supported by major manufacturers such as QSC, Audinate, Ravenna, and Merging Technologies. It forms the base layer for many modern AoIP ecosystems, often operating in tandem with proprietary protocols like Dante and Ravenna. However, despite broad compatibility, AES67 adoption has been somewhat fragmented because implementation details vary slightly among vendors, and end‑user configuration can be complex. The future will be defined by efforts to simplify deployment and to make AES67 a truly plug‑and‑play experience. Some networks still require manual PTP domain configuration and stream subscription, but new tools and device profiles are rapidly automating these steps.

Key Drivers for Future Adoption

Standardization and Interoperability Demands

Broadcasters and studios increasingly operate hybrid environments that mix legacy analog/digital gear with IP‑based equipment. AES67’s ability to bridge different protocols is a primary driver. As more facilities migrate to fully networked workflows, the demand for a universal audio transport layer will intensify. Industry bodies such as the AES and the JT‑NM (Joint Taskforce for Networked Media) are actively refining AES67 profiles to ensure consistent behavior across products. This standardization effort means that in the near future, a broadcast engineer will be able to plug any AES67‑compliant device into a network and have it synchronize automatically—without manual configuration of PTP or stream parameters. New “plug-and-play” implementations based on mDNS and DNS‑SD are already appearing, reducing the need for static stream addressing.

Remote Production and Cloud Integration

The shift toward remote production—accelerated by the pandemic—remains a strong trend. AES67 is uniquely suited to support distributed workflows because it operates over standard IP networks and can be tunneled across the internet with appropriate bandwidth and latency management. Future AES67 implementations will likely incorporate more robust error correction and jitter buffers, making it feasible to transport high‑channel‑count audio across wide area networks with the same reliability as local networks. Cloud‑based production platforms, such as those from Sony, Grass Valley, and Lawo, are already integrating AES67 to allow seamless audio routing from remote stages to centralized mixing consoles. As cloud adoption grows, AES67 will become the de facto standard for interconnecting on‑premises and cloud‑based audio resources. Emerging virtualized audio routers, running on commodity servers, natively speak AES67, enabling remote contribution workflows where talent can send multichannel audio from a home studio to a cloud mixer over a simple VPN.

Integration with SMPTE ST 2110 and Dante

SMPTE ST 2110: The Video‑over‑IP Companion

AES67 is the audio component of the SMPTE ST 2110 suite of standards for professional media over managed IP networks. While ST 2110‑30 (Audio) directly references AES67, future developments will see even tighter coupling. Broadcasters currently building ST 2110 plants require a video+audio transport that operates with sample‑accurate timing. AES67’s use of PTP (IEEE 1588) for synchronization aligns perfectly with ST 2110’s timing model. We can expect future AES67 profiles to support additional audio formats (e.g., 192 kHz sampling rates, higher bit depths) to match the capabilities of ST 2110 video streams. Moreover, the adoption of AES67 for contribution links—where high‑channel‑count audio accompanies uncompressed video—will become more common as network bandwidth costs decrease. The new SMPTE ST 2110‑22 standard for compressed video will also benefit from AES67’s flexible audio routing, allowing broadcasters to carry both uncompressed and compressed audio streams within the same infrastructure.

Dante and Proprietary Protocol Integration

Audinate’s Dante remains the most widely deployed AoIP protocol in the pro‑audio market. While Dante uses its own network discovery and timing mechanisms, it can transparently bridge to AES67 streams via the “Dante Domain Manager” or through native support on newer Dante hardware. In the future, we can anticipate a convergence where Dante devices automatically expose AES67 streams for interoperability without a separate configuration step. Similarly, Ravenna, an open standard derived from AES67, will see deeper integration with both Dante and ST 2110. The outcome will be a single, unified IP audio fabric where engineers can route signals from a Dante microphone array to an AES67‑native mixing console and then to an ST 2110 broadcast channel—all without format conversion or signal degradation. Some manufacturers are already shipping hybrid interfaces that can simultaneously operate in Dante and AES67 mode, allowing legacy Dante networks to connect seamlessly to ST 2110 installations.

Enhanced Technical Capabilities

Latency Management and Redundancy

One of the early criticisms of AES67 was its relatively high baseline latency compared to proprietary solutions. However, ongoing refinements—such as reduced packet size options (48 samples per packet instead of 1024) and improved PTP profiles—will bring AES67 latencies well under 1 millisecond in a local network. For live production, this is critical for monitoring foldback and immersive audio workflows. Future implementations will also embrace redundancy mechanisms like stream switching (active/passive) and network diversity, similar to what ST 2022‑7 provides for video. This will make AES67‑based audio networks as robust as their SDI counterparts, encouraging risk‑averse broadcasters to commit fully to IP. Hitless seamless protection switching, where the receiver automatically selects the best available stream, is being added to the AES67 specification through the new “AES67 Redundancy” profile.

Higher Channel Counts and Immersive Audio

As object‑based audio and immersive formats (such as Dolby Atmos and MPEG‑H) become mainstream in broadcast, the number of audio channels per program is increasing dramatically. AES67’s capability to carry up to 256 channels per flow (depending on bit depth and sample rate) will be extended in future revisions to support uncompressed 512‑channel streams. This will enable broadcasters to deliver fully immersive commentary, multilingual tracks, and metadata alongside the main audio bed. The standard will also evolve to support transport of audio metadata and spatial audio extensions natively, rather than relying on proprietary side‑channels. For instance, future AES67 payloads may include embedded ADM (Audio Definition Model) metadata, allowing the receiving equipment to render object-based audio automatically.

Enhanced Synchronization and Timing

Accurate sample‑level synchronization is vital for multi‑microphone setups, remote contributions, and video‑audio alignment. AES67 uses PTP version 2 (IEEE 1588‑2008) to achieve submicrosecond time alignment. Future AES67 profiles will likely adopt the more precise “default” PTP profile used by ST 2110, eliminating the need for separate PTP domains for audio and video. Moreover, we may see support for “hybrid PTP” that works seamlessly across local networks and wide areas, enabling distributed studios to mix audio from multiple remote locations without phasing issues. Boundary clocks and transparent clocks are becoming standard in network switches designed for media, further increasing timing accuracy over long-haul links.

Forward Error Correction and Packet Loss Recovery

Although AES67 was originally designed for managed local networks where packet loss is near zero, the move to remote production over the public internet requires resilient transport. Emerging extensions to AES67 incorporate Forward Error Correction (FEC) techniques, such as Reed-Solomon coding, to reconstruct lost packets without retransmission. These mechanisms will allow AES67 to operate reliably over connections with up to 5% packet loss, making it viable for cloud and WAN deployments. The AES is also evaluating the use of the FEC framework defined in SMPTE ST 2022-7 for audio streams, which will standardize an interoperable approach across vendors.

Market Expansion and Ecosystem Growth

From High‑End to Mid‑Market and Education

The cost of AES67‑enabled devices has been steadily declining. Small‑format mixing consoles, Dante‑to‑AES67 bridges, and USB audio interfaces now routinely offer AES67 support at price points that were once unthinkable. This trend will accelerate as silicon manufacturers embed AES67 compatibility in standard networking chips. By 2028, virtually all professional audio interfaces—including inexpensive micro‑array recorders—will include built‑in AES67 connectivity, just as they currently include Ethernet. The result will be a dramatic expansion of the total addressable market: educational institutions, houses of worship, corporate AV systems, and small production houses will all adopt AES67 as a low‑cost, high‑flexibility solution. The release of the AES67-2023 revision which clarified several implementation details, has already encouraged more chipset vendors to add native support.

Manufacturer Support and Third‑Party Certification

Major manufacturers are investing heavily in AES67 compliance. Audinate has already certified many Dante devices as “AES67‑compatible,” and the list is growing monthly. Companies like Cisco, Arista, and Netgear are optimizing their network switches for AES67 traffic, with pre‑configured QoS profiles. The future will see an AES67 certification program analogous to Dante Certification, offering tiers for network engineers, system designers, and operators. This ecosystem growth reduces the risk for end users, as they can source interoperable gear from a wide range of brands without worrying about vendor lock‑in. The Audio Engineering Society is working with the AVnu Alliance to develop an official AES67 interoperability badge, which will appear on compliant products from 2025 onward.

Role of Open Source and Community Standards

The open‑source community is also contributing to AES67 adoption. Projects like “AES67‑light” (a reference implementation) and the “Linux Audio Over IP” initiative provide building blocks for custom deployments. As these tools mature, integrators will be able to build AES67 endpoints on low‑cost single‑board computers, further reducing entry barriers. This democratization of AoIP will empower small innovators to develop niche products—such as wireless microphone receivers with built‑in AES67 output—that were previously only available from large vendors. The open-source ecosystem also promotes rapid prototyping: a field engineer can deploy a Raspberry Pi running Linux with AES67 support as a temporary stream gateway during live events.

Training and Certification Initiatives

Industry‑Wide Education Programs

The successful adoption of any technology depends on the knowledge of the workforce. Recognizing this, the AES Education Committee has launched online courses and workshops specifically focused on AES67 networking, PTP configuration, and stream management. Major trade shows like NAB Show, IBC, and AES Convention now feature dedicated AES67 training sessions. Over the next few years, we can expect these programs to become formalised into a recognised certification—much like the “Certified Audio Engineer – AoIP” credential. Such certifications will give employers confidence that their staff can design, deploy, and troubleshoot large‑scale AES67 networks. Manufacturers like Lawo and Riedel also offer in‑depth training on their AES67‑based ecosystems, often integrated with broader IP media training.

Hands‑On Training and Remote Labs

In addition to online learning, manufacturers are offering hands‑on training tools. For example, Audinate’s “Dante Virtual Soundcard” and the free “AES67 Connection Manager” allow students to experiment with AoIP routing without expensive hardware. Universities are building “IP audio labs” where students can build practice networks that mirror real‑world broadcast plants. The combination of affordable hardware and accessible training curricula will produce a new generation of engineers who are fluent in IP audio from day one of their careers. The Audinate AES67 resource center provides downloadable lab exercises and example configurations that educators are incorporating into their syllabuses.

Challenges and Considerations for the Future

Despite the positive outlook, several challenges must be addressed to achieve universal adoption. One key issue is network complexity: while AES67 itself is a standard, the underlying IP network must be properly configured—with QoS, VLANs, and PTP domains—to deliver reliable audio. Without adequate training, network misconfiguration can lead to dropouts or high latency. Another challenge is interoperability testing: even though products claim AES67 compatibility, subtle differences in implementation can cause problems. The industry needs a robust, vendor‑neutral interoperability testing program, similar to the “AVnu Alliance” certification for AVB. The AES67 Plugfest events have shown progress, but a formal certification regime is still pending. Additionally, security becomes a concern when AES67 streams traverse wide area networks or the public internet. Current AES67 lacks encryption and authentication, making it vulnerable to eavesdropping and spoofing. Future versions of the standard will likely include optional encryption (e.g., AES-256) and authentication mechanisms, possibly adopted from the SMPTE ST 2110 security framework. Finally, bandwidth and latency trade‑offs on congested networks require careful planning; while 1 Gbps is sufficient for most studio needs, larger installations may require 10 GbE backbones. The adoption of software‑defined networking (SDN) and automated provisioning tools can help manage these requirements, but the industry must invest in them.

Conclusion

The future of AES67 in broadcast and studio environments is bright. As the standard matures, we can expect deeper integration with video‑over‑IP standards like SMPTE ST 2110, simplified configuration through unified PTP domains, and dramatically lower costs that bring AES67 to all tiers of the audio industry. Enhanced technical capabilities—including ultra‑low latency, higher channel counts, native support for immersive audio, and built‑in forward error correction—will enable production workflows that were previously impossible. At the same time, expanded training initiatives and manufacturer certification programs will ensure that the workforce can harness these capabilities effectively. AES67 is not merely a passing standard; it is evolving into the foundational plumbing for all professional audio‑over‑IP systems. Broadcasters, studio owners, and audio professionals who invest in AES67‑based infrastructure today will be well‑positioned to benefit from the industry’s inevitable shift toward fully networked, flexible, and scalable audio environments. For more information on the AES67 standard itself, visit the AES official standard page. For practical deployment guides, consult the Audinate AES67 resource center. To see how AES67 fits into the wider AoIP landscape, the SMPTE ST 2110 suite provides essential context. The era of fully interoperable audio‑over‑IP is here, and AES67 is leading the way.