StormAudio - Knowledge Center

AoIP option - Output configuration

Step 1 - ISP configuration

By default the ISP output sampling rate is configured at 48 kHz. In case the stream you wish to connect to runs at different sampling rate, follow the steps outlined in https://support-stormaudio.atlassian.net/wiki/spaces/SUP/pages/309592160/AoIP+option+-+Input+configuration#Settings-definition.

Note: the AoIP option can only manage a single sampling rate on both input and output, either 48 kHz or 96 kHz.

Step 2 - Connect the ISP destination

 

Case 1: AES67/Ravenna devices

ANEMAN - Configuration

Based on Ravenna/AES67 protocol, the AoIP option module is designed to work with the ANEMAN software to manage connections between compatible AES67 devices.

Consult this link to download the ANEMAN software and its manual.
To troubleshoot ANEMAN Issues, see ANEMAN User Manual Troubleshooting section.

We recommend downloading ANEMAN plugins for compatibility with all devices.

Open “Manage plugins” in the “Action” tab to access the plugin manager menu. Click the “GET IT” button of the plugins to install them.

Launch ANEMAN. You should see a Discovery zone, a Devices view and a Matrix view:

Devices recognized by ANEMAN should appear in the “WORLD ZONE”. Here is an example of an AES67 device connected to the same network as the PC:

If one of these views is not shown in ANEMAN, click on the “View” menu to enable them.

Click on the “New Sample Rate Zone”. ANEMAN will create a new zone where the sampling rate is the same for all devices inside it. Simply drag and drop the devices you want to link together in the sampling rate zone.

Set the master of this zone by dragging and dropping the device on the crown icon :

You can access the WebUI -if it exists- of all devices by right clicking on its icon in the World / Pinned view.

Access any device’s WebUI with a right click, “Web Services”:

  • AoIP option name: Basic WebUI

  • Advanced page: Full settings WebUI

  • Maintenance page: WebUI to update the firmware

Connect an AES67 destination

When devices are detected, they appear in the ANEMAN discovery zone. Selecting them will populate the matrix view on the right :

This matrix is used to create connections between sources and destinations. It will automatically generate the Session sources and sinks accordingly in the AoIP option WebUI.

After clicking on a cell of the matrix, click on “Apply Multicast” or “Apply Unicast”:

  • Multicast allows multiple devices to be connected to this source.

  • Unicast restricts to only one device connected to this source.

ANEMAN will now make the connections between the devices, the connections will turn Purple to confirm the connections are established and working (will turn Blue for Unicast connections).

Removing a connection:

  • Select one or multiple connections by clicking on it in the matrix. Then click on “Delete connections” to remove it.

Hover your cursor over a connection to get more information about it in the matrix view.

The ISP can manage both input and output streams at the same time.

Case 2: Dante devices

This section explains how to configure an ISP and a third party Dante device as a destination.

Notes:

  • Dante devices must be AES67 capable.

  • Dante Virtual Sound card is not AES67 capable.

  • Dante devices currently only support 48 kHz in AES67 mode.

  • Dante in AES67 mode can handle a maximum of 32 streams.

First, connect all of the following network devices to the same network switch :

  • An ISP equipped with an AoIP option.

  • An AES67 capable Dante device.

  • A computer with Dante Controller installed.

Dante Controller - Configuration

Most devices first require that you enable their AES67 mode, either from Dante Controller or from their specific remote access application.

Enable AES67 mode in Dante Controller:

  • Go in the Devices > Device View page.

  • Select your Dante device.

  • Go in the AES67 Config tab.

  • Set the AES67 mode to Enabled. A reboot is required if it was disabled.

  • Make sure the Multicast Address Prefix is set to 239.IP.xxx.xxx. “IP” must be the same for all devices on the AoIP network, default is 69.

  • Go now in the “Device Config” tab. Set the Latency to 2 or 5 ms (AES67 recommended value is 3 ms).

For a stereo or latency critical configuration with few channels, you can select 2 ms.

For theaters with multiple channels, select 5 ms to ensure the streams' stability.

Connect to a Dante destination

  • Enter the AoIP option WebUI, either with ANEMAN or by clicking on the AoIP option IP in the “settings” tab.

  • In the Session sources tab, create a session source (transmitter) using the icon at the upper left corner.

  • Select the desired IO module if necessary.

  • Make sure the multicast address ("Address" field) is using the Dante multicast prefix. Default is 239.69.xxx.xx.

  • Set the Channel count to 8 and select the channels you need to stream below.

 

  • Open the Dante Controller software.

  • Connect Transmitters to Receivers as desired. See a classic mapping below:

Cisco SG350 Users

For users wanting to connect AES67 Dante devices to their Cisco SG350 switches, please note that an additional Multicast Group configuration must be done for every port used by these devices.

Dante AES67 users have to add IP Multicast Groups:

Once the configuration file has been applied, and the switch has been rebooted:

  • Connect to the Cisco Administration page (default address with the Merging configuration file is 169.254.1.254).

  • Make sure the Display Mode of the Cisco Administration page is set to Advanced.

  • Browse to Multicast>IP Multicast Group Address and click on Add.

  • Enter VLAN ID 1 (assuming you have only 1 VLAN) and enter 224.0.0.230 as IP Multicast Group Address. Click on Apply.

  • Now select the 224.0.0.230 Group and click on Details. Set the ports connected to Dante devices to Static and click on Apply.

  • Repeat the same operation for addresses 224.0.0.231, 224.0.0.232 and 224.0.0.233.


Related pages

https://support-stormaudio.atlassian.net/wiki/pages/resumedraft.action?draftId=309133371

StormAudio - Knowledge Center