Intercom Call Configuration
- 31 Mar 2025
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Intercom Call Configuration
- Updated on 31 Mar 2025
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IP Call Configuration
An IP call is a direct call between two intercom devices using their IP addresses, without a server or a PBX. IP calls work when the devices are on the same network.
Make IP Calls
Make IP calls by pressing the Dial key on the home screen, entering the IP number such as “192✳168✳35✳123”, and pressing the Call button.
IP Call Configuration
Enable IP call on the Phone > Call Feature > Direct IP interface.
Direct IP Port: Set the port for direct IP calls. The default is 5060, with a range from 1-65535. If you enter a value within this range other than 5060, ensure consistency with the corresponding device for data transmission.
SIP Call Configuration
Session Initiation Protocol(SIP) is a signaling transmission protocol used for initiating, maintaining, and terminating calls.
A SIP call uses SIP to send and receive data between SIP devices, and can use the internet or a local network to offer high-quality and secure communication. Initiating a SIP call requires a SIP account, a SIP address for each device, and configuring SIP settings on the devices.
SIP Account Registration
Each device needs a SIP account to make and receive SIP calls.
Akuvox intercom devices support the configuration of two SIP accounts, which can be registered under two independent servers.
Click here to view the SIP account registration example.
Register the SIP account on the Account > Basic interface.
Status: Indicate whether the SIP account is registered or not.
Account 1/Account 2: The door phone supports 2 SIP accounts.
- Account 1 is the default account for call processing. Also, it will be utilized when the Akuvox SmartPlus cloud service is activated.
- The system switches to Account 2 if Account 1 is not registered.
- To designate the account to be used for outgoing calls, select the account number for contacts or dial plan prefixes in their settings.
Tip
For configuring contact call and dial plan, see here.
When the device is connected to the SmartPlus Cloud, the display label, register name, and username will show its SIP number.
Display Label: The label of the device.
Display Name: The designation for Account 1 or 2 is to be shown on the device itself on the calling screen.
Register Name: Same as the username from the PBX server.
User Name: Same as the username from the PBX server for authentication.
Password: Same as the password from the PBX server for authentication.
SIP Server Configuration
SIP servers enable devices to establish and manage call sessions with other intercom devices using the SIP protocol. They can be third-party servers or built-in PBX in Akuvox indoor monitor.
To set it up, go to the web Account > Basic interface.
Server IP: Enter the server’s IP address or its domain name.
Port: Specify the SIP server port for data transmission.
Registration Period: Define the time limit for SIP account registration. Automatic re-registration will initiate if the account registration fails within this specified period.
SIP Call DND&Return Code Configuration
The Do Not Disturb(DND) feature prevents unwanted incoming SIP calls, ensuring uninterrupted focus. It also allows you to set a code to be sent to the SIP server when rejecting a call.
Set it up on the web Phone > Call Feature > DND interface.
Account: Select the account that applies DND.
Return Code When DND: Specify the code sent to the caller via the SIP server when rejecting an incoming call in DND mode.
DND On Code: Turn on the DND on the server using the code obtained. The DND On Code is 78 by default.
DND Off Code: Turn off the DND on the server using the code obtained. The DND Off Code is 79 by default.
Outbound Proxy Server
An outbound proxy server is used to receive all initiating request messages and route them to the designated SIP server in order to establish a call session via port-based data transmission.
Set it up on the Account > Basic > Outbound Proxy Server interface.
Server IP: Enter the SIP proxy IP address.
Port: Set the port for establishing a call session via the outbound proxy server.
Backup Server IP: Enter the SIP proxy IP address to be used when the main proxy malfunctions.
Port: Set the proxy port for establishing a call session via the backup outbound proxy server.
Data Transmission Type
Akuvox intercom devices support four data transmission protocols: User Datagram Protocol(UDP), Transmission Control Protocol(TCP), Transport Layer Security(TLS), and DNS-SRV.
Set it up on the Account > Basic > Transport Type interface.
UDP: An unreliable but very efficient transport layer protocol. It is the default transport protocol.
TCP: A less efficient but reliable transport layer protocol.
TLS: An encrypted and secured transport layer protocol. Select this option if you wish to encrypt the SIP messages for enhanced security or if the other party’s server uses TLS. To use it, you need to upload certificates for authentication. To use it, you need to upload certificates for authentication.
DNS-SRV: A DNS service record defines the location of servers. This record includes the hostname and port number of the server, as well as the priority and weight values that determine the order and frequency of using the server.
SIP Hacking Protection
Internet phone eavesdropping is a network attack that allows unauthorized parties to intercept and access the content of the communication sessions between intercom users. This can expose sensitive and confidential information to the attackers. SIP hacking protection is a technique that secures SIP calls from being compromised on the Internet.
Set it up on the Account > Advanced > Call interface.
Prevent SIP Hacking: Activate this feature to only receive calls from contacts in the whitelist. This protects users’ private and secret information from potential hackers during SIP calls.
Call Session Timer
SIP does not have a built-in way to keep track of active sessions. While the user agent can figure out if a session has timed out using its own specific method, the proxy server lacks this capability. As a result, the proxy server may not always know if a session is still ongoing. For instance, if a user agent fails to send a BYE message at the end of a session or if the BYE message gets lost due to network issues, the proxy server won't be aware that the session has ended. Consequently, the proxy server will continue to hold the call status without knowing if it's still valid.
To address this problem, RFC4028 introduces a survival mechanism for SIP sessions. Either the user agent or the proxy server periodically sends re-INVITE or UPDATE requests to keep the session active. The interval between these update requests is determined by the negotiation mechanism defined in the session. If no update request is received within the specified interval, the session is considered terminated.
Set it up on the web Account > Advanced > Session Timer interface.
Active: The call session timer is disabled by default.
Session Expire: Enter the session call duration before the call expires or ends automatically for refreshment. For example, if you set the session expiration as 1800 seconds (Ranging from 90- 7200 sec), you can have the door phone terminate the ongoing call with another intercom device in 1800 seconds.
Session Refresher: Select UAC (User Agent Client) or UAS (User Agent Server) for the call session refreshment.
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