Audio Intercom User Guide

NOTE: Audio Intercom functionality is available on Rithum Switch Pro hardware only with firmware release 4.0+.

Current Capabilities:

• Receive a SIP audio call from an intercom entry point to Rithum Switch Pro. Calls can be:
  • Peer-2-Peer (P2P); or
  • via registration with a SIP server (firmware 4.4+)
• Send DTMF unlock code within a call

Limitations:

• Audio-only, video support is available via an additional plugin
• SIP transport is via UDP or TCP, TLS not supported
• Audio media is via RTP, SRTP not supported
• Ethernet only. Note that whilst Rithum Switch Pro has Wi-Fi, Video Streaming and Audio Intercom features are not officially supported over Wi-Fi. That said, they may well work suitably.

Tested With

This has been tested with the following third party solutions. Please let us know if we can add more systems to this list.

• 2N IP Verso (& Control4 DS2) - See the 2N user guide here
• Doorbird - See the Doorbird user guide here
• Ubiquiti UniFi Access
• Akuvox
• Linphone
• BAS-IP AV-06
• Fasttel

Configuration:

P2P SIP Calls

In many intercoms, it is possible for a button push to initiate multiple P2P SIP calls simultaneously, thereby allowing the intercom to ring on multiple Rithum Switch Pros.

The intercom unit will need to be configured 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 SIP user/extension 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.

Example screenshots below are for configuration of a 2N IP Verso.

If possible, set a 'Display Name' for the SIP account in the intercom unit. This name will then be displayed as the caller on the Rithum Switch intercom screen. For example:

Configure 'Intercom' settings on Rithum Switch Pro:

• 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
• SIP Registered: If registration with a SIP server is configured then this item shows the current registration state
• SIP Configuration: If SIP server registration is required then follow the link shown to go to the SIP server configuration web page (or scroll up to reveal the QR code). Refer to the SIP server configuration section below for further details

SIP Server Configuration (firmware 4.4+)

The intercom unit will need to be configured to make a call to the registered user/extension of the switch, e.g.

sip:1003@<Server IP address/domain>:5060

Visit http://<switch IP address/settings/sipaccount.html to open the SIP server configuration web page as shown below.

NOTE: This settings page is accessible on the network unless a PIN is set for the settings screen (System > Set PIN), which also protects this settings page.

• SIP Registered: This shows the current SIP registration state and updates every few seconds. There may be a delay before this changes to "Yes" after saving settings
• Enable Registration: This enables/disables registration with the account details provided. Note that it's possible to set a "Display Name" for use with P2P calls without registration being enabled
• User/Extension: The SIP account username to register with. In some PBX-style SIP servers this will likely be an extension number
• Password: The password for the SIP account user/extension being registered with
• Domain (optional): The SIP server domain. This will often be the same as the server address and if omitted then 'Server Address' will be used
• Server Address: The URL or IP address of the SIP server. Note that the port number should not be appended to the address, it is provided separately in the field below
• Server Port: The port number to be appended to the SIP server address. For the SIP protocol this is normally 5060
• Display Name (optional): A display name to be shown on the intercom unit when a call is active to this switch

Video Streaming

It is also possible to add a video stream to the SIP call and access a video stream from scene button. To do that you require the Video Streaming Plugin and can read the guide here: Video Streaming User Guide