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Authentication ID: Enter the username/ID that is used by the PBX registrar to authenticate the
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Video Intercom. This field cannot be left empty.
Password: Enter the password that is used to authenticate the Video Intercom. This field is
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optional and can be left empty.
Domain: Enter the domain name that the Video Intercom will register to. Typically this will be
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the same as the SIP server, but if your SIP server handles multiple domains, this will need to be
confirmed. This must be a valid IPv4, IPv6, or DNS name.
Registration Duration: Enter the time period, in seconds, that the Video Intercom will wait
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before re-registering with the SIP server. This must be a value greater than 0.
Tip: The Registration Status field shows the current registration status of the Video
Intercom. If the Video Intercom is not successfully registered with the SIP server, the
status will display the reason for unsuccessful registration.
PBX Call Destination: Enter the extension number that the Video Intercom will use when a
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visitor makes a call. This field cannot be left empty.
5. Click Apply to save your changes.
When a SIP phone user receives a call, they can answer the call and speak to the visitor. If the visitor is
approved to enter, the user can use their DTMF code to grant door access. For more information, see
Configuring DTMF Codes on the next
SIP Network Settings
Configure SIP Network Settings to facilitate voice intercom and other communication on networks using the
same SIP network settings.
1. Select SIP as the Intercom Call Destination.
2. Select the PBX mode checkbox.
3. Modify the following SIP Network Settings or use the default values:
SIP Port: Enter the SIP port number. Must be a valid port number between 1 and 65534. Cannot
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be the same port number as the RTP Base Port or any other camera ports assigned to the
Video Intercom.
RTP Base Port: Enter the RTP base port number. Must be a valid port number between 1 and
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65534. Cannot be the same port number as the SIP Port or any other camera ports assigned to
the Video Intercom. The RTP Base Port is the first port in a 2-port range, so make sure the input
RTP Base Port and RTP Base Port+1 are not already in use.
DTMF payload number: Enter the DTMF payload number.
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Opus codec payload number: Enter the Opus codec payload number.
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H264 codec payload number: Enter the H264 codec payload number.
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4. Click Apply to save your changes.
SIP Network Settings
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