3.2.3.1 Phone


Phone is an essential function of 2N® Indoor Talk allowing you to establish connections to other IP network terminals. 2N® Indoor Talk supports the extended SIP.


Parameter List

2N® Indoor Talk Phone includes the following four tabs:

  • SIP 1,2 – complete SIP account settings.
  • Calls – incoming and outgoing call settings.
  • Audio – audio codec, DTMF and other audio stream parameter transmission settings.
  • LAN device – general parameters and count of identified LAN devices.

SIP 1 and SIP 2

Two SIP accounts can be configured on 2N® Indoor Talk.

  • SIP ACcount Enable – allow the SIP account use for calling. If disallowed, the account cannot be used for making outgoing calls and receiving incoming calls.



  • Display name – set the name to be displayed on the called user's phone as CLIP.
  • Phone number (ID) – set your device phone number (or another unique ID composed of characters and digits). Together with the domain, this number uniquely identifies the devices in calls and registration.
  • Domain – set the domain name of the service with which the device is registered. Typically, it is identical with the SIP Proxy or Registrar address.
  • Test call – display a dialogue window to make a test call to the selected phone number, see below. 



  • Authentication ID – enter the alternative user ID for the device authentication. Phone Number (ID) will be used if this parameter is left empty.
  • Password – set the authentication password. If your PBX requires no authentication, the parameter will not be applied.



  • Proxy Address – set the SIP Proxy IP address or domain name.
  • Proxy Port – set the SIP Proxy port (typically 5060).
  • Backup Proxy Address – set the backup SIP Proxy IP address or domain name to be used if the main Proxy fails to respond to requests.
  • Backup Proxy Port – set the backup SIP Proxy port (typically 5060).



  • Registration Enabled – enable device registration with the set SIP Registrar.
  • Registrar Address – set the SIP Registrar IP address or domain name.
  • Registrar Port – set the SIP Registrar port (typically 5060).
  • Backup Registrar Address – set the backup SIP Registrar IP address or domain name to be used if the main Proxy fails to respond to requests.
  • Backup Registrar Port – set the backup SIP Registrar port (typically 5060).
  • Registration Expires – set the registration expiry, which affects the network and SIP Registrar load by periodically sent registration requests. The SIP Registrar can alter the value without letting you know.
  • Registration State – display the current registration state (Unregistered, Registering..., Registered, Registration terminated...).
  • Failure Reason – display the cause of failure of the last registration attempt – the Registrar's last error response is displayed, e.g. 404 Not Found.


  • SIP Transport Protocol – set the SIP communication protocol: UDP (default), TCP or TLS.
  • Lowest Allowed TLS Version – set the lowest TLS version to be accepted for device connection.
  • Verify Server Certificate – verify the SIP server public certificate against the CA certificates uploaded in the device.
  • Client Certificate – specify the client certificate and private key used for verifying the intercom’s authority to communicate with the SIP server.
  • Local SIP Port – set the local port for thedevice for SIP signaling. A change of this parameter will not be applied until the deviceis restarted. The default value is 5060.
  • PRACK Enabled – enable the PRACK method for reliable confirmation of SIP messages with codes 101–199.
  • REFER Enabled – enable call forwarding via the REFER method.
  • Send Keep Alive Packets – set that the device shall inquire periodically about the state of the called station via SIP OPTIONS requests during the call (used for station failure detection during the call).
  • IP Address Filter Enabled – enable the blocking of SIP packet receiving from addresses other than SIP Proxy and SIP Registrar. The primary purpose of the function is to enhance communication security and eliminate unauthorized phone calls.
  • Receive encrypted calls only (SRTP) – set that SRTP encrypted calls shall only be received on this account. Unencrypted calls will be rejected. At the same time, TLS is recommended as the SIP transport protocol for higher security. 
  • Encrypted outgoing calls (SRTP) – set that outgoing calls shall be SRTP encrypted on this account. At the same time, TLS is recommended as the SIP transport protocol for higher security. 
  • Use MKI in SRTP Packets – enable the use of MKI (Master Key Identifier) if required by the counterparty for master key identification when multiple keys rotate in the SRTP packets.
  • Do Not Play Incoming Early Media – disable playing of the incoming audio stream before the call sent by some PBXs or other devices is picked up (early media). A standard local ringtone is played instead.
  • QoS DSCP Value – set the SIP packet priority in the network. The set value is sent in the TOS (Type of Service) field in the IP packet header. Enter the value as a decimal number. A change of this parameter will not be applied until the deviceis restarted. 
  • External IP Address – set the public IP address or router name to which the device is connected. If the device IP address is public, leave this parameter empty.
  • Starting RTP Port – set the initial local RTP port in the range of 64 ports used for audio and video transmission. The default value is 4900 (i.e. the range is 4900–4963). The parameter is only set for account 1 but applies to both the SIP accounts.
  • RTP Timeout – set the audio stream RTP packet receiving timeout during a call. If this limit is exceeded (RTP packets are not delivered), the call will be terminated by the device. Enter 0 to disable this parameter. The parameter is only set for account 1 but applies to both the SIP accounts.
  • Broadsoft Compatibility Mode – set the Broadsoft PBX compatibility mode. Having received re-invite from a PBX in this mode, the intercom replies by repeating the last sent SDP with currently used codecs instead of sending a complete offer.
  • Rotate SRV Records – allow SRV record rotation for SIP Proxy and Registrar. This is an alternative method of transition to backup servers in the event of main server failure or unavailability.



Calls



  • General Settings – set the call duration limit after which the call is automatically terminated. The intercom signals termination with a beep 10 s before the call end. Enter any DTMF character into the call (# on your IP phone, e.g.) to extend the call time.



  • Call Answering Mode (SIP1, SIP2) – set the way of receiving incoming calls. The following three options are available: 
    • Always busy – the device rejects incoming calls. 
    • Manual answering – the device signals incoming calls by ringing and the user can answer the call using a numeric keypad. 
    • Automatic – the device picks up incoming calls automatically. You can set the call receiving mode for each SIP account separately.
  • Local Call Receiving Mode – set the way of receiving incoming local calls. The following three options are available: 
    • Always busy – the device rejects incoming calls. 
    • Manual Pickup – the device rings to signal incoming calls and the user can press a keypad button to pick up.
    • Automatic – the device picks up incoming calls automatically. You can set the call receiving mode for each SIP account separately.
  • Pick up in – this parameter is only active when the Automatic parameter pickup mode is enabled. The call is picked up automatically after the preset timeout.
  • Time Limited DND Mode – Disable ringing for a limited time (silent mode). If this function is activated, the device shall not ring for a limited time when a call comes in.
  • Reject Calls in DND Mode – if this function is activated, the device reject calls in the Do not Disturb mode. The function can be used for immediate call redirection at absence to a mobile phone call, for example.
  • DND Mode for Doorbell Button – if this function is activated, the device shall not ring when the doorbell button is pressed.
  • Do Not Disturb Mode with Time Profile – hoose one or more time profiles to be applied. Set the time profiles in Directory / Time profiles.
    • – select one of the pre-defined profiles or set the time profile for the given element manually.


  • Connecting Time Limit – set the maximum outgoing call connection timeout after which the calls are automatically terminated. If the calls are routed to the GSM network via GSM gateways, you are advised to set a value higher than 20 s.
  • Ringing Time Limit – set the maximum call setup and ringing time in which all outgoing calls are automatically terminated. If the calls are routed to the GSM network via GSM gateways, you are advised to set a value longer than 20 s. Minimum value: 1 s, maximum value: 600 s. Set 0 to disable the time parameter.

Audio


  • Enable/disable the audio codecs offered for call setup and set their priorities. 

The tab below helps you define how DTMF characters shall be sent from the intercom. Check the DTMF receiving options and settings of the opponent to make the function work properly.


  • In-Band (Audio) – enable the classic method of sending DTMF in the audio band using standardised dual tones.
  • RTP (RFC-2833) – enable sending DTMF via the RTP according to RFC-2833.
  • SIP INFO (RFC-2976) – enable sending DTMF via SIP INFO messages according to RFC-2976.

The tab below helps you define how DTMF characters shall be received from the intercom. Check the DTMF receiving options and settings of the opponent to make the function work properly.


  • In-Band (Audio) – enable classic DTMF dual tone receiving in the audio band.
  • RTP (RFC-2833) – enable DTMF receiving via the RTP according to RFC-2833.
  • SIP INFO (RFC-2976) – enable DTMF receiving via SIP INFO messages according to RFC-2976.



  • QoS DSCP value – set the audio RTP packet priority in the network. The set value is sent in the TOS (Type of Service) field in the IP packet header.
  • Jitter Compensation – set the buffer length for compensation of interval unevenness in audio packet arrivals. Set a higher value to increase the receiving immunity at the cost of a higher sound delay. 

Local Calls



  • Enable Local Calls – enable calls between devices in the LAN. With this function off, the other LAN devices cannotlocate this device, i.e. cannot call the device in the device:device_ID format.



  • Device ID – Set the device ID to be displayed in the LAN device list in all the 2N devices in one and the same LAN. You can direct a call to this device by setting the user phone number as device:device_ID in these devices.



  • Access Key 1, 2 – set the access key shared by the 2N answering units and intercoms. If the keys in the 2N answering units and intercoms fail to match, the device cannot communicate, i.e. the intercom cannot call the 2N answering unit and vice versa.



  • Access Key – set the access key shared by the 2N answering units. If the keys in the 2N answering units fail to match, the answering units cannot communicate, i.e. the answering units cannot call each other.
  • Multicast Address – set the network multicast address to which the anwering unit message shall be sent.



  • LAN Device count – display the number of local devices in the network.
  • Show LAN devices list – display a detailed list of local devices in the network.