Setting up an AES67-compliant audio network in your studio transforms how you route, synchronize, and scale audio signals. Unlike traditional analog or proprietary digital systems, an AES67 network gives you the freedom to mix and match equipment from different manufacturers while maintaining sub-millisecond latency and sample-accurate synchronization. This guide covers every step in detail — from initial planning and switch selection to device configuration, PTP timing, and ongoing management — so you can build a production-ready audio-over-IP infrastructure that meets the AES67 standard.

What Is AES67 and Why It Matters for Your Studio

AES67 is a technical standard developed by the Audio Engineering Society that defines interoperability for audio-over-IP (AoIP) networks. It specifies coding, stream synchronization, and network transport parameters so that devices from different vendors — like a Focusrite RedNet interface, a Neumann MT 48 monitor controller, and a Yamaha CL5 console — can all share audio channels over a single Ethernet cable. Before AES67, studios were often locked into proprietary ecosystems (e.g., Dante, Ravenna, Livewire) that required costly bridges or complex gateways to interconnect. AES67 broke down those walls by creating a common language for commercial AoIP.

The practical benefits for your studio include:

  • Vendor flexibility – buy the best mic preamps, converters, and DSP units without worrying about protocol compatibility.
  • Scalability – add more devices, rooms, or venues by simply connecting another switch.
  • Low latency – typical AES67 networks achieve under 1 ms round-trip latency with proper configuration.
  • Sample-accurate synchronization – using Precision Time Protocol (PTPv2/IEEE 1588-2008), all devices stay locked to a master clock, eliminating drift across long cable runs.

If your studio already uses Dante, Ravenna, or Livewire, those protocols often have an AES67 compatibility mode. However, a pure AES67‑compliant network ensures the highest degree of interoperability and future‑proofing. For an overview of the standard, visit the AES67-2018 official document page.

Prerequisites and Equipment Checklist

Building an AES67 network requires more than just audio gear. Here’s a full list of what you’ll need before you start cabling:

Network Hardware

  • Managed Gigabit Ethernet switches – at least two for redundancy, or one for smaller studios. Switches must support IEEE 802.1Q VLANs, IGMP snooping (to prevent multicast flooding), and most importantly, IEEE 1588 PTPv2 boundary or transparent clock capabilities. For production studios, consider models from Cisco Catalyst 9300 series, Netgear M4300 series (specifically the AV line), or TP-Link JetStream. Avoid unmanaged switches entirely.
  • Cat6a or Cat7 shielded twisted‑pair cables – AES67 can operate over copper for runs up to 100 m. For longer distances, use fiber transceivers (SFP+ modules) and single‑mode or multimode fiber.
  • Optional: PTP grandmaster clock – while any device can serve as the PTP master, a dedicated grandmaster (e.g., from Meinberg, Ardenna, or Focusrite) delivers the highest stability, especially for multi‑room or broadcast applications.

Audio Devices and Software

  • AES67‑compatible audio interfaces, consoles, or DSP units. Check the manufacturer’s documentation for AES67 compliance. Many Ravenna devices (e.g., Merging Technologies Horus, DirectOut .sb) are native AES67. Dante devices can enable AES67 mode through the Dante Controller.
  • Network management software – typically each manufacturer provides its own app (e.g., Dante Controller, Ravenna Manager, Livewire Session Manager). You may also use industry‑wide tools like Wireshark for deep packet inspection.
  • Basic knowledge of IP networking: subnet masks, DHCP vs static IPs, and multicast addresses.

Step 1: Network Planning and Topology Design

Skipping the planning phase is the most common cause of AES67 headaches. Begin by drawing a physical diagram of your studio: locate every audio device, switch, and computer. Mark the distances between them — anything beyond 100 m requires fiber.

Choice of IP Addressing Scheme

Use static IP addresses for all AES67 devices. DHCP can cause address changes mid‑session, breaking audio streams. Choose a private subnet (e.g., 10.0.0.0/8 or 192.168.100.0/24) that doesn’t overlap with your general office network. A /24 subnet gives you 254 addresses — plenty for most studios. Avoid the 169.254.x.x link‑local range (APIPA) as it can interfere with PTP.

VLANs: Isolating Audio Traffic

Configure a dedicated VLAN for all audio traffic (e.g., VLAN 100). This keeps multicast streams from interfering with file transfers, internet bound traffic, and control signals. On your managed switches, assign each audio device’s port to the audio VLAN and set the PVID accordingly. Ensure that the switch port connecting to your computer or DAW is either also in the audio VLAN or that you route through an access port — but never mix audio and data on the same broadcast domain.

Also configure an access port for the management computer (or a dedicated management VLAN) so you can control devices without flooding your audio network with ARP or DHCP requests. For complex multi‑room studios, use separate VLANs per room and route them through a layer‑3 core switch or router.

IGMP Snooping and Multicast Management

AES67 streams use IP multicast (typically in the 239.0.0.0/8 range). Without IGMP snooping, every multicast packet is sent to all switch ports, increasing traffic and causing dropped packets. Enable IGMP snooping on all managed switches. Optionally, use IGMP querier to maintain multicast groups when no router is present. Some switches also allow you to set per‑port multicast bandwidth limits — useful for preventing a rogue stream from saturating the pipe.

Step 2: Configuring Network Switches for AES67

Once the topology is planned, physically connect all equipment. Then configure each switch. The exact commands vary by manufacturer, but the principles are universal:

  1. Disable Energy Efficient Ethernet (EEE) – EEE powers down transceivers during idle periods, causing PTP timing jitter.
  2. Enable jumbo frames (MTU 1500 or larger) – AES67 typically uses 48, 96, or 192 sample packet sizes; jumbo frames help reduce overhead. Set MTU to at least 1500, ideally 9000 (nine thousand) bytes.
  3. Set Spanning Tree Protocol (STP) to Rapid‑PVST+ or RSTP – STP is necessary for loop prevention, but ensure it’s not blocking audio ports. Use edge ports for devices that only connect to one switch.
  4. Activate PTP boundary clock mode – if your switch supports it. This reduces PTP timing accumulation across many hops. For switches without PTP hardware support, ensure that PTP packets are not delayed or queued differently (e.g., set strict priority for PTP traffic).
  5. Apply Quality of Service (QoS) – mark PTP (DSCP AF11 or CS7) and audio streams (DSCP EF) as high priority. Use a strict priority queue on egress ports.

If you are using a managed switch without PTP boundary clock capabilities, make sure to configure an extra‑tight latency budget. You can find detailed switch recommendations in the Audinate Dante technical documentation, which applies to many AES67 setups.

Step 3: Device Configuration and Clock Synchronization

Now configure each audio device. Power everything on and access each unit’s web interface or control panel. Here’s the checklist:

Assigning Static IPs

Set a unique IP address, subnet mask, and default gateway for every device. Record these in a spreadsheet for future reference. For devices that support multiple network ports, ensure the audio port is configured on the audio VLAN.

Enabling AES67 Mode

Many modern interfaces come with AES67 as a mode inside a proprietary framework. For example, in Dante Controller, right‑click the device and select Device View → Audio Settings. Look for the AES67 checkbox. In Ravenna Manager, simply choose AES67 stream format. On Livewire devices, set the protocol to AES67 in the front‑panel menu.

If your device only supports Ravenna or Dante without AES67, check that the firmware is up to date. Most manufacturers added AES67 support in recent releases.

Setting Up PTP (IEEE 1588) Synchronization

All AES67 networks must use PTPv2. One device acts as the grandmaster. Typically, you choose a dedicated PTP grandmaster or a high‑quality audio interface as the clock source. Here’s how to configure:

  • On the grandmaster: select PTP profile for AES67 (default profile: domain 0, two‑step, 1‑second sync interval). Enable “only announce if priority criteria met.”
  • On all slave devices: select “PTP for synchronization” and set the same domain number (usually 0).
  • Check that each device reports the grandmaster’s identity. For fine‑tuning, adjust the announce interval to 2 seconds and sync interval to 0.5 seconds.
  • If you have a dedicated Meinberg or Ardenna grandmaster, it will automatically select the best clock in the network.

PTP works best in a star topology with switches that support boundary or transparent clocks. If you experience timing jitter, reduce the number of switch hops between the grandmaster and the farthest device. For a deeper dive into PTP for professional audio, refer to the IEEE 1588 overview.

Step 4: Creating Audio Streams and Routing

With the network stable and all devices synchronized, you can start sending audio. AES67 uses the Session Description Protocol (SDP) to define streams. Each stream is a unicast or multicast flow of one channel or multiple channels. In most control software, you create a “transmitter” and a “receiver” (or “sender” and “subscriber”).

Using Control Software

  • Dante Controller: Route audio by dragging a transmitter’s channel to a receiver’s channel. For AES67 streams, use the “AES67” tab to view and connect SDP‑based streams.
  • Ravenna Manager: Right‑click a device → Add Stream → select multicast address, sample rate, number of channels. Then subscribe on the receiving device.
  • Livewire Session Manager: Create sources and destinations; assign multicast IPs manually (e.g., 239.1.10.1).

When setting up multicast streams, choose unique multicast addresses in the range 239.0.0.0–239.255.255.255. Avoid conflicts with other devices. For unicast, use the receiving device’s IP address.

Latency and Packet Size

AES67 defines three latency profiles: 0.125 ms, 1 ms, and 2 ms (based on packet size of 6, 48, or 96 samples at 48 kHz). The default profile is 1 ms (48 samples). Use 0.125 ms only on a dedicated, low‑latency network with PTP boundary clocks. For most studios, 1 ms is safe and gives plenty of headroom.

Step 5: Testing the Network

Before you rely on the system for a critical session, run a battery of tests.

Connectivity and Ping Tests

From your computer, ping every device by its IP address. Ensure less than 1 ms round‑trip time (RTT) over a single switch. If you see RTTs above 2 ms, check for switch bottlenecks or faulty cables. Also verify that the latency is consistent — wild fluctuations indicate network congestion or misconfigured QoS.

PTP Synchronization Verification

Most AES67 control software displays clock offset. On Dante Controller, look at the Clock Status pane. A healthy network shows offsets under 1 microsecond (0.000001 seconds) between master and slaves. If offsets are high, adjust PTP priority values or check switch PTP capabilities.

Audio Stream Integrity

Play a known test tone (e.g., 1 kHz sine wave at –18 dBFS) from a transmitter and measure it at the receiver with an analyzer like REW or SignalScope. Confirm the signal appears at the correct level and phase. Check for crackles, dropouts, or silence. For multi‑channel streams, verify channel mapping.

Stress Testing

Simulate peak traffic by streaming all planned channels simultaneously. Monitor CPU load on switches and devices. Use Wireshark to capture a few seconds of traffic and filter for AES67 (L24, L16) or RTSP streams. Look for retransmissions — they indicate packet loss. If you see more than 0.01% loss, re‑evaluate network design.

Step 6: Ongoing Management and Troubleshooting

Once the system is running, keep an eye on these areas:

  • Switch firmware updates – new releases often improve PTP stability and multicast handling.
  • VLAN isolation – never plug an unmanaged switch into an AES67 port; it will break VLAN segmentation.
  • Clock master redundancy – program a secondary grandmaster (e.g., a backup device with higher PTP priority 2). If the primary fails, the network should auto‑switch without dropping audio.
  • Temperature and power – hot switches cause clock drift. Ensure adequate ventilation in your equipment rack.

Common Issues and Fixes

  • No audio stream visible: Verify that AES67 mode is enabled on both sender and receiver. Check that the multicast address isn’t blocked by IGMP filtering. Try changing to unicast.
  • Audio clicks or pops: Usually timing related. Confirm PTP sync is within tolerance. Increase packet size to 2 ms (96 samples) temporarily. If that fixes it, you have a network delay issue — look at QoS or switch hops.
  • Stream shows but no sound: Check sample rate and bit depth match (e.g., 48 kHz/24‑bit). AES67 mandates 24‑bit but some devices allow 16‑bit. Also verify channel count alignment.
  • High latency on DAW monitoring: Ensure your computer’s network interface is not in power‑saving mode. Disable all energy‑efficient Ethernet settings in the NIC driver.

Conclusion

A properly configured AES67 network delivers the reliability and flexibility that modern studios demand. By planning your topology, using managed switches with PTP support, and carefully synchronizing every device, you can transport dozens of channels with sub‑millisecond latency and pristine clarity. The steps outlined above — from VLAN isolation to stream creation and stress testing — form a repeatable workflow that works for everything from a single control room to a multi‑facility broadcast center. As your studio grows, the same modular network can expand without starting from scratch, proving that AES67 is not just a standard, but a strategic investment in your audio infrastructure.