StormAudio - Knowledge Center

AoIP option - Input configuration

Step 1: ISP configuration

Input definition

First you need to define the AoIP module as a source in the ISP:

  • Go to the “Inputs” tab of the ISP’s WebUI.

  • Click on the “N” button in front of the “16 ch AES In” input line to enable AoIP as source. You can rename this input as you wish, “16 ch AES In” is the default name given to our 16ch AES/EBU input optional module.

  • Select the video source associated to the AoIP stream and “32ch AES67” as Main audio in path:

This new input becomes available in the remote control and can be selected :

Settings definition

Then, go to the “Settings” tab and check the AoIP settings available under Input/Output Setup list:

  • AES67 Frequency:

The global network sampling rate can be set to 48 or 96 kHz. The ISP AoIP sampling rate must be the same as the other AoIP devices on the network.

  • AES67 Module hostname:

The name of the AoIP option in your ISP. It can be changed in its WebUI. See “AES67 Module IP Address” to access it.

  • AES67 Module IP Address:

Click on it to access the AoIP module WebUI.

  • AES In processed:

Set this to “On” to allow the source to be upmixed according to the selected preferred upmixer. In this case, the input is limited to 8ch with a fixed mapping on the ISP side that should be replicated in the source device.

  • Edit 32 Channels digital input mapping:

Allows you to manage the 32 inputs mapping manually. It is possible to connect each input to a particular speaker type:

The incoming AoIP stream does not provide information on its channel mapping. Therefore, you need to manually map the incoming signals to the corresponding speakers in your theater using the signals available in the dropdown list, in the “New Mapping” area.

The incoming AoIP stream might not match your Theater speaker configuration perfectly. In case there is no matching signal in the stream, leave the according speaker signal to OFF. It will remain silent in AoIP mode unless StormXT is enabled.

Step 2: Connect the source to the ISP

Case 1: AES67 / Ravenna devices


Based on AES67/Ravenna 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”:

  • In the World Zone, every device recognized by ANEMAN should appear. 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 :

Every device gives access to its WebUI if it exists 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 source

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 allow multiple devices to be connected to this source.

  • Unicast restrict 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.

Case 2 : Dolby Cinema Media Server/Processor

Dolby Media Servers and processors are able to decode Dolby streams and send them through your AoIP network. They are typical AoIP inputs for your ISP.

In this section, we’ll cover the CP850, CP950 and IMS3000 Cinema Media Servers/processors.

Dolby devices features

  • The CP950 is a 7.1 processor, able to send 8 AoIP channels (or more with Dolby Atmos option). The CP850 and IMS3000 are natively able to manage more channels.

  • The Dolby Atmos Designer tool allows management of the signal mapping for each stream.

  • The CP850 cannot be the PTP slave. The Merging devices can be set as PTP slave, but then can't be locked to external sync (such as WordClock or video ref).

  • The CP850 does not allow devices and sources discovery, and neither provides a SDP retrieving mechanism. Therefore, ANEMAN cannot be used to discover the device and the streams connections must be done manually.

Before configuring a CP850, note that this device implements a particular version of AES67, which generates conflicts with some devices on the same network. Dante devices has compatibility issues with CP850 in the same network.

See section below with IMS3000 and CP950 which support any kind of other device on the network.

Connect a Dolby source to your ISP’s AoIP

Dolby Cinema Processors require two network connections, one for the control and one for the AoIP/AES67 network.

  • First connect the COMMAND port to your data network. This network configuration can be accessed on the unit front display, and can be set to DHCP or static IP, depending on your needs. Please refer to the Dolby documentation for further details. The command network is only required to configure the device, the network cable can be removed afterwards.

  • Connect the DOLBY ATMOS CONNECT OUT to your AoIP/AES67 network.

CP850 - CP950 - IMS3000

This section will cover the CP850 configuration, the process is very similarfor all 3 devices.

  • Open the CP850 web-app in your browser; type in the url search bar its Command IP address, then enter the login credentials (see CP850 documentation).

  • Go in System > Preferences and set the Dolby Atmos Connect Protocol to AES67.

  • Go to System > Network and configure the Dolby Atmos Input.

Set the IP configuration to manual, and enter a valid IP address and Netmask.  (Gateway can be left to

  • Disable jumbo frames option should be ticked in. Then click on Apply.

  • In our example we will use the address and a mask.

  • Now switch to the Dolby Atmos Connect tab. Make sure legacy mode option is unticked.

  • Set the Static Source IP (=> CP850 AES67 IP address).

  • PTP Domain Number should be set to 0, and both PTP priorities set to 100. This will set the CP850 as PTP GrandMaster.

  • Destination multicast IP : the destination device multicast IP address. It should be , you can use what you want. In our example is chosen.

  • Source UDP Ports (1-8, 9-16, 17-24,....): 6517, 6519, 6521,...
    RTP destination UDP ports (1-8, 9-16, 17-24,....): 6518, 6520, 6522, .... They must be even numbers.

  • Then click on Apply.

  • Set the ISP device sampling rate to 48 kHz through ISP WebUI in the setting tab. (96kHz not supported).

  • Open your ISP device advanced page in your browser.

  • Set the Latency to AES67 (48 smpl).

  • In the PTP tab, Status section, please note the Master GMID (not the one in the Master section !). This value will be required later.

  • Go now to the Session Sinks tab.

  • Press on the Create session sink button.

  • Set your required output in the IO drop down menu.

  • Set the Channel count to 8 and click on Apply.

  • Label the connection, in our example we will label it CP850_1

  • Download our SDP example and open it in a Text editor (notepad or notepad++).
    You will now have to adapt it to your network configuration.


no modification


Enter your Dolby Atmos Connect IP (Point 5 in the CP850 Configuration section above)
In our example:
o=- 1 0 IN IP4


Enter your Sink Label, in our example :

c=IN IP4 239.1.XXX.XXX

XXX Enter the Destination Multicast IP you entered in the Dolby Atmos Connect page 
(Point 7 in the CP850 Configuration section above), in our example :
c=IN IP4

a=clock-domain:PTPv2 0

No modification


RTPPORTDESTINATION is the RTP Destination UDP Port defined in the CP850.
(Point 8 in the CP850 Configuration section above)
In our example :
m=audio 6518 RTP/AVP 96  for channels 1 to 8
m=audio 6520 RTP/AVP 96  for channels 9 to 16

a=rtpmap:96 L24/48000/8

No modification


MASTERGMID is the PTPMaster ID, that we have noted previously in the
Advanced pages > PTP tab > Status section.
In our example :


No modification

  • Now that the SDP is configured for your network, go back in the Advanced pages > Sync Tab.

  • Click on the Manual checkbox, and copy - paste the SDP configuration.


  • Click on Apply to confirm the configuration

  • You will now have to map this connection :


  • The CP850 is now linked to the ISP AoIP input.

  • You may now remove the CP850 Command network connection.

Additional remarks

If you need to add more channels (for tracks 9-16, 17-24,....), please modify the RTP Destination UDP Port,
as configured in the CP850 Dolby Atmos Connect Page.
In our example : m=audio 6518 RTP/AVP 96  for channels 1 to 8 / m=audio 6520 RTP/AVP 96  for channels 9 to 16 / ....

PTP Domain : 

You may use another PTP domain than 0.  In such a case you will have to set the domain number in all the devices manually.
You must also modify the SDP description with the correct domain number: 
a=clock-domain:PTPv2 DomainNumber => a=clock-domain:PTPv2 109 if you want to use PTP Domain 109
a=ts-refclk:ptp=IEEE1588-2008:MASTERGMID:DomainNumber => in our example a=ts-refclk:ptp=IEEE1588-2008:00-D0-46-FF-FF-02-C2-B6:109

Case 3: Dante devices

This section explains how to configure an ISP with a third party Dante device for both inputs and outputs.


  • 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 “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 msec).

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

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

Connect a Dante source

In Dante Controller :

  • Go in the Devices > Device View page.

  • In the Devices drop down menu, select Create Multicast Flow.

  • Tick the AES67 Flow box, and select the channels you need to transmit.

  • Then click on Create.

  • Open the AoIP option WebUI advanced page, either via ANEMAN or by clicking on its IP in “setting” Tab.

  • On the Session sinks tab, create a session sink using the icon at the upper left corner.

  • Select the desired IO module if necessary.

  • Click on the arrow next to the "Source" field, you will get a list of the currently available SAP sources.

  • Select your Dante stream.

  • The stream will now connect and the connection icon in the left column will turn green.

  • Depending on the device version, a RTP Status will display the current state of the stream. Receiving packets indicates a good connection.

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StormAudio - Knowledge Center