The Case for AES67 in Modern Podcast Studios

Podcast production demands pristine audio quality and seamless connectivity across an expanding array of sources. Traditional analog or digital point-to-point wiring quickly becomes brittle as studios add microphones, mixers, recorders, and remote guests. Each new device often requires its own dedicated cable run, patchbay configuration, and format converter, creating a tangle of high-cost, hard-to-maintain infrastructure. Audio-over-IP (AoIP) networks solve these scalability and flexibility issues by replacing point-to-point connections with a single, managed Ethernet network that carries multiple audio streams simultaneously. The AES67 interoperability standard is central to making AoIP practical across diverse hardware ecosystems. By adopting AES67, podcast studios gain a vendor-neutral foundation that simplifies integration, reduces latency, and future‑proofs the infrastructure against proprietary lock‑in. Whether you’re building a small home studio or a multi-room production hub, understanding AES67 is a critical step toward a more adaptable, high-performance audio setup.

What Is AES67?

AES67 is a technical standard published by the Audio Engineering Society (AES). It defines a common way to transport high‑quality digital audio over IP networks using RTP (Real‑time Transport Protocol) over UDP. The standard specifies sampling rates up to 96 kHz and bit depths of 16 or 24 bits, supporting up to eight audio channels per stream. Key to its value is interoperability: AES67 allows devices built for different proprietary AoIP systems (Dante, Ravenna, Livewire, Q‑Sys) to discover each other and exchange audio streams without a gateway or format converter. This removes vendor lock‑in, letting studio owners choose the best hardware for each task while maintaining a unified network.

AES67 was first released in 2013 and has been updated to align with broadcast standards such as SMPTE ST 2110-30. It relies on Precision Time Protocol (PTPv2, IEEE 1588-2008) for clock synchronization, ensuring that all devices sample at the same instant and that audio remains phase‑coherent across the network. The standard also defines multicast addressing and stream discovery, making it possible to route audio from a microphone preamp to a mixing console or recorder with a few clicks in management software. Each stream is described by a Session Description Protocol (SDP) file, which contains the multicast IP address, port number, sample rate, bit depth, and other parameters. Receiving devices download this SDP to lock onto the stream, making configuration largely automated when using management tools that support AES67.

Key Benefits of AES67 for Podcast Studios

1. Interoperability That Cuts Complexity

The most immediate advantage of AES67 is the ability to mix devices from different ecosystems. A podcast studio might use a Dante‑enabled microphone preamp, a Ravenna‑based digital mixer, and a recorder that supports Livewire. With AES67 all three can communicate on the same switched Ethernet network. This eliminates the need for multiple converter boxes or dedicated audio snakes. For a growing studio, that means lower upfront cost, simpler cabling, and straightforward expansion. Instead of being locked into a single brand’s product line, you can choose the best-in-class microphone preamp from one manufacturer, the most intuitive console from another, and the most reliable recorder from a third — all working together transparently.

2. Improved Flexibility and Scalability

Because AES67 runs over standard Ethernet, studio layouts become software‑defined. You can relocate a microphone or change the routing from a control pad without pulling new cables. Adding a new room for interview recording? Just drop a managed switch and configure the streams. Remote guests can be brought in over secure WAN connections using AES67‑enabled encoding, and the same network can carry control data alongside audio if proper QoS (Quality of Service) is in place. This agility is especially valuable in podcast production, where show formats frequently change — one day you might need a three-person roundtable, the next a solo interview with a remote guest. AES67 lets you reconfigure the studio in minutes rather than hours.

3. Low Latency for Live Production

Podcasting often involves live segments where host and guest interact in real time. AES67 specifies a maximum latency of 1 ms per hop under ideal conditions, and practical deployments achieve sub‑5 ms end‑to‑end across a small studio. That is significantly lower than many proprietary AoIP systems in their default modes. Combined with PTP‑based synchronization, AES67 ensures that multiple microphone signals arrive at the mixer without phase cancellation or drift – essential for natural‑sounding conversations. Even when adding a remote guest via a codec and WAN link, careful AES67 configuration can keep total latency under 20 ms, which is imperceptible to listeners.

4. Future‑Proofing and Broadcast Compatibility

Broadcast and post‑production are increasingly adopting ST 2110, a suite of standards that uses AES67 as its audio transport layer. Any podcast studio that intends to produce content for television or streaming services will benefit from being AES67‑ready. It also positions the studio to adopt higher‑channel‑count systems, audio‑over‑IP monitor solutions, and even audio‑follows‑video workflows without a rip‑and‑replace rebuild. As the professional audio industry moves away from legacy TDM (time-division multiplexing) and MADI, AES67 provides a migration path that protects your investment in microphones, consoles, and outboard gear.

How to Implement AES67 in a Podcast Studio

Step 1: Audit Your Existing Equipment

Begin by checking whether your current hardware supports AES67 natively. Many professional audio interfaces and mixing consoles from manufacturers like Focusrite, Yamaha, Allen & Heath, and RME include AES67 support alongside Dante or Ravenna. If a device only supports a proprietary protocol, verify that its firmware can be updated to enable AES67, or budget for a media converter. For example, devices that natively support Dante can usually be configured to stream out AES67 by changing a multicast mode setting in the Dante Controller software. Some older devices may require a firmware upgrade — always check the manufacturer’s support site for the latest AES67 compliance notes.

Step 2: Build a Robust Network Infrastructure

AES67 requires a managed Ethernet network with full‑duplex, switched architecture. Use Gigabit Ethernet minimum (10 GbE recommended for studios with many streams). Enable IGMP snooping and querier to prevent multicast traffic from flooding ports. Configure DiffServ (DSCP) QoS to prioritize audio packets, marking them with the EF (Expedited Forwarding) class recommended by the AES67 standard. Ensure all switches support PTPv2 (IEEE 1588-2008) with boundary clock or transparent clock capabilities – consumer‑grade switches often lack this, leading to clock drift and dropped audio.

A typical small podcast studio can run on a single managed switch, but for larger setups spanning multiple rooms, use spanning tree protocol (RSTP) to prevent loops and redundant links. Keep the network dedicated to audio if possible; sharing it with heavy data traffic can introduce latency spikes. When selecting switches, look for models from vendors like Netgear, Cisco, or Aruba that explicitly mention AES67 or PTP boundary clock support. A good starting point is a 24-port Gigabit managed switch with PoE+ capability for powering microphones and intercoms.

Step 3: Configure Clocking and Synchronization

All AES67 devices must synchronize to a single grandmaster clock. Choose one device (often a master clock generator or a console) as the PTP grandmaster. In Dante systems, this is usually the device with the highest clock priority. In Ravenna environments, the master is typically the audio interface or a dedicated clock source. Most modern managed switches can act as PTP boundary clocks, reducing jitter in larger networks.

Connect all audio devices to the same PTP domain. Set the sample rate and bit depth uniformly across the network – 48 kHz / 24‑bit is a common choice that balances quality and bandwidth. If you need to mix different sample rates, use SRC (sample rate conversion) at the network edge, but this adds latency. Use the device’s web interface or dedicated software (e.g., Dante Controller, Ravenna Manager) to confirm that all devices are locked to the grandmaster with a PTP offset of less than 1 microsecond.

Step 4: Create and Route Streams

Use the device’s control software to configure AES67 streams. For example, in the Dante environment, you create a multicast transmit flow that outputs audio from a specific channel and assign it as an AES67 stream. In Ravenna, use the manufacturer’s web interface to set the stream format, destination multicast address, and session description protocol (SDP) file. The SDP file provides the information that receiving devices need to lock onto the stream. Some audio consoles can automatically discover AES67 streams via mDNS or SAP announcements.

Test routing end‑to‑end: send a 1 kHz tone from a microphone preamp to the mixer and verify that the level and phase are correct. Confirm that the stream appears in the receiver’s “AES67 Inputs” list. In larger networks, assign static multicast addresses to avoid conflicts with other services. Document each stream’s multicast IP and SDP file location for future reference.

Step 5: Perform Interoperability and Latency Testing

Even with correct configuration, real‑world interoperability issues can arise. Test every device combination with a known audio source. Measure latency using a loop‑back test: send a signal from a known low‑latency device (e.g., a microphone with direct monitor) and compare the arrival time on the mixer. Acceptable end‑to‑end latency for podcasting is under 10 ms; typical AES67 implementations achieve 1–3 ms. If latency is higher, check PTP synchronization status and ensure no switch is introducing large buffer delays. Also verify that the audio stream format matches the receiver’s allowed channels and bit depth – mismatches are a common cause of silence or distortion. Use Wireshark with an AES67 display filter (rtp.payload.audio) to inspect packet arrival times and identify any jitter or loss.

Comparing AES67 with Other AoIP Protocols

Dante

Audinate’s Dante is the most popular AoIP protocol in pro audio. It is proprietary and uses a combination of PTP and its own clocking. Dante devices can be configured to output AES67 streams, but not all Dante features (redundancy, unicast) are available in AES67 mode. For a podcast studio that already owns Dante gear, enabling AES67 is a low‑cost way to gain interoperability, but you lose some Dante‑specific conveniences. Dante offers a rich ecosystem of controllers and software, and its primary advantage is plug‑and‑play simplicity. If you’re building a studio from scratch and don’t need to integrate with third‑party gear, Dante alone may be sufficient. However, AES67 opens the door to devices from Ravenna and Livewire ecosystems.

Ravenna

Ravenna is an open technology developed by ALC Network and used by Lawo, Neumann, and others. It is closely aligned with AES67 and is essentially a superset that adds redundancy and higher channel density. Ravenna devices work natively with AES67 without configuration changes. For studios planning to scale to broadcast‑style routing, Ravenna is a strong choice. Its support for ST 2022-7 seamless redundancy makes it ideal for mission‑critical live productions. Ravenna’s network management tools are more advanced, but they also require a steeper learning curve. Podcast studios already using Neumann microphones or Lawo consoles will find Ravenna to be the most natural path.

Livewire

Livewire (from Telos Alliance) is another AoIP system common in radio and podcast production. Its newest version, Livewire+, fully supports AES67, allowing Axia consoles and Pathfinder routing to interoperate with Dante or Ravenna gear. This is especially valuable in studios that mix podcasting with live broadcast. Telos Alliance provides extensive documentation on integrating Livewire+ with AES67, including sample SDP files and recommended QoS settings. Many radio stations have adopted Livewire as their backbone, and podcasters who collaborate with radio networks will benefit from this compatibility.

By adopting AES67 as a common denominator, studios can combine hardware from these ecosystems. A producer might use a Dante microphone preamp, a Ravenna‑based DAW, and a Livewire+ talkback system – all on the same network – without any conversion hardware beyond Ethernet switches. This vendor‑agnostic approach is the cornerstone of modern, future‑proof studio design.

Practical Considerations for Podcast Studios

Network Redundancy

For mission‑critical live podcasting, consider implementing ST 2022-7 seamless redundancy. This standard allows two independent network paths for each AES67 stream. If one switch fails or a cable is cut, the receiving device picks up the other stream with zero audio glitch. Many Ravenna and Dante devices support this. For a smaller studio, a single managed switch with uninterruptible power supply may be sufficient. However, if you’re producing live shows that cannot afford any dropouts, investing in a redundant network topology with dual switches and redundant PTP grandmasters is wise.

Remote Guests and WAN Connectivity

AES67 is designed for local area networks, but with careful engineering it can work over VPN or dedicated links. Use lower sample rates (44.1 kHz) to reduce bandwidth, and implement forward error correction. Alternatively, use dedicated AoIP codecs that wrap AES67 streams for secure internet transport. For simple remote guest calls, many podcasters prefer low‑latency software solutions (e.g., Cleanfeed) and then bring the audio back into the AES67 network via an analog or USB bridge. If you frequently record remote guests, consider a purpose‑built encoder from manufacturers like Digigram or Comrex that supports AES67 over WAN.

Power Over Ethernet (PoE)

AES67 devices sometimes support PoE, which simplifies cable management for microphones, intercom panels, and small mixers. Ensure your switch provides sufficient PoE budget (802.3af/at). However, note that PoE adds complexity to network planning as power draw must be calculated per port. For example, a Neumann MT 48 audio interface draws about 15W, and a typical 24-port PoE+ switch provides up to 370W total. Plan your budget accordingly, and consider using PoE injectors for high‑power devices.

Firewalls and Security

AES67 traffic uses UDP ports in a range from 1024 to 65535, depending on the device. For multicast, IGMP must be allowed. Some network security policies block such traffic. In a dedicated audio network, configure permissive firewall rules; if the audio network shares infrastructure with corporate IT, work with the IT team to create VLANs that isolate audio traffic while allowing password‑protected web access for configuration. Always change default passwords on switches and audio devices to prevent unauthorized access. While AES67 itself has no built‑in encryption, placing the audio network on its own VLAN provides a layer of isolation from general traffic.

Case Studies: Real-World AES67 Deployments

Berlin Podcast Network – A mid‑sized podcast network in Berlin upgraded from an analog patchbay to a fully AES67‑enabled setup. They replaced three separate systems (one Dante‑based microphone rack, a Ravenna digital mixer, and a Livewire+ coder) with a single gigabit network. Each control room has a small managed switch; the core network uses a 10 Gb backbone with PTP boundary clock support. The result: they can reconfigure routing for any of six studios in under a minute, and they added a remote interview suite that connects via a dedicated MPLS link. Latency measured under 3 ms end‑to‑end. The cost of the network infrastructure was recovered within one year by eliminating the need for dedicated digital snakes and format converters.

University Podcast Lab – A university department converted a traditional radio studio into a 12‑seat podcast lab using AES67. Each seat has a Dante‑enabled microphone interface and a Ravenna‑compatible headphone amp. A single Cisco Catalyst switch manages all streams, with PTP grandmaster provided by an external LTC clock. Students can create up to three separate podcast channels simultaneously, each with independent mixing and recording. The lab’s instructor can monitor any stream from a central Ravenna controller. The system has been used for over two years with zero audio drops, and the department plans to expand to a second lab using the same AES67 backbone.

Troubleshooting Common AES67 Issues

  1. No Audio Received – Verify that the multicast stream is announced and that receivers are in the same PTP domain. Check that the SDP file matches exactly on sender and receiver. Use Wireshark with an AES67 decoding display filter (rtp.payload.audio) to confirm packets are reaching the receiver’s IP. Also ensure that the receiver’s clock is synchronized to the grandmaster — a PTP offset greater than 1,000 ns can cause stream rejection.
  2. Intermittent Audio / Dropouts – Usually caused by network congestion. Examine switch statistics for lost packets. Increase DSCP priority for audio traffic (set to EF = 46). Reduce the number of hops between devices. Ensure PTP grandmaster clock is stable and all switches support boundary clock. Enable IGMP snooping to prevent multicast traffic from flooding non‑subscribing ports.
  3. High Latency – Check that no device is introducing internal buffer delays. Some interfaces have a “low latency” mode that should be enabled. Also ensure that the network’s PTP profile matches the AES67 requirements (profile AE01). Mismatched PTP profiles can cause clock drift that appears as latency variation. Use a loop‑back test with a known signal to isolate the delay source.
  4. Clock Drift – If audio starts in sync but gradually loses alignment, the grandmaster clock may be of low quality. Use an external GNSS‑derived clock or a master clock generator with a stability of better than 1 ppm. Verify that all switches pass PTP transparently — some unmanaged switches strip or corrupt PTP timestamps. Check PTP log messages on each device to see if it is failing to synchronize.

Conclusion

AES67 is not just an interoperability standard – it is a practical enabler for flexible, scalable, and future‑ready podcast studios. By replacing a web of proprietary cables and converters with a single managed Ethernet network, studio owners simplify operation, reduce costs, and open the door to new workflows. The key to success lies in careful network design: choose managed switches that support PTP and IGMP, configure QoS correctly, and test thoroughly. As more manufacturers embed AES67 natively, the barrier to entry continues to drop. Whether you are building a two‑person podcast booth or a multi‑room production hub, AES67 provides a solid foundation for high-quality audio connectivity and flexibility. For further reading, consult the AES Standards Committee, check out Audinate’s AES67 guide, explore ALC Network’s Ravenna resources, and review Telos Alliance’s Livewire+ AES67 documentation for deeper technical details and configuration examples.