Control4 DS2 / DS3 Configuration for Audio Intercom

In order to make a call from a Control4 DS2/DS3 intercom to a Rithum Switch Pro, the intercom needs to be configured to make a P2P SIP call.

Since you will likely be integrating the intercom both with Control4 and with Rithum, in the intercom settings you need to configure two SIP servers: one for Control4, the other for Rithum. Note you don't actually need a SIP server for the Rithum side, that's just how the intercom is configured.

To do this, follow the steps below. The exact steps may differ slightly depending on updates to the web interface, but the principles are the same.

The configuration is done in the 2N interface rather than in Control4. You should configure the DS with Control4 first before you follow the steps below.

1. Configure SIP2

a) Navigate to SIP 2 and check SIP Account Enabled. Here you will want to enter a Display Name which will be shown on the incoming call notification and call active screens on Rithum. The Phone Number (ID) and Domain are not necessary and something arbitrary can be entered if they can't be let blank. Alternatively, use the Router IP for the Domain.

b) Leave SIP Registrar Registration disabled, since we will be using a P2P SIP call, and not registering to a SIP server. Set SIP Transport Protocol to UDP or TCP, TLS not supported, and the Local SIP Port to 

c) Configure Audio Codecs & DTMF

Configure the audio codecs as required, but the below settings show what is supported and recommended. DTMF Sending can be configured as desired and does not serve a purpose for this integration. For DTMF Receiving (i.e. receiving to the intercom from Rithum) both RTP and SIP INFO are supported and can be configured as desired.

2. Configure User(s)

a) In the 2N web interface, navigate to the Directory and Add a user. This user can be for a single Rithum Switch Pro, or you can configure it to call multiple switches simultaneously (up to three plus a deputy user), as you'll see below.

Additionally, you can configure a 2N button push to call one or more users (up to 16), so in this way you can define multiple Rithum Switch Pros to be called from each button push.

b) Scroll down to the User Phone Numbers section. In the first Phone Number field, enter the address to make a direct P2P SIP call to the switch, the address/phone number to enter being:

sip:<Switch IP address>:5060

Some systems may require a caller ID to be added, in which case you can use anything generic such as 123 in the following example:

sip:123@<Switch IP address>:5060

Note: It is recommended to bind the IP address of each of your Rithum Switch Pros within your router settings so that they don't change.

Importantly, you want to select SIP2 instead of SIP1 (which is what Control4 is using). This will append a /2 to the end of the phone number.

c) If you are setting this user up as multiple switches to, you can add additional addresses in the Phone Number field(s) for Number 2 and/or Number 3. Most likely you would want to enable Group Call to Next Number so that the switches are called simultaneously.

2. Configure Button Calling

Navigate to Calling and Dialing. Here you can configure which user(s) your button(s) call by selecting them.

[NB: If your interface looks different to this, please update the intercom firmware]

4. Configure Relay(s)

a) This will likely vary depending on where you are wiring your release, but as an example, navigate to Hardware > Switches. Check Switch Enabled and configure the Output Settings and Switch Control as desired.

Under Activation Codes enter the Code you want to use to release the lock/door (e.g. 00) and set the accessibility to DTMF Only.

b) Configure Rithum Switch Pro to send the correct code in the correct DTMF Mode as per Audio Intercom settings

• Initiate Demo Call: Press this to initiate a demo call which will cause the intercom screen to be shown
• Unlock DTMF Code: Enter DTMF digits to be sent when the intercom screen unlock button is pressed. The default code for 2N is 00*
• DTMF mode: Select RTP or SIP INFO depending on what options are available in the intercom unit
• Hang up after unlock: Enable this option to automatically end the intercom call when the unlock button is pressed

5. Video Streaming

It is also possible to add a video stream to the SIP call and access DS video streams from scene buttons. To do that you require the Video Streaming Plugin and can read the guide here: Video Streaming User Guide. Note that you may require a 2N Enhanced Video licence to enable RTSP streaming.