The Phone service is one of the basic functions of the device 2N^® Clip – helps you establish connections with other IP network terminal equipment. The device supports the extended SIP.
The Phone menu includes 6 folders:

SIP 1 and SIP 2– complete SIP account settings.
Calls – incoming and outgoing call settings.
Audio – audio codec, DTMF and other audio stream parameter transmission settings.
Video – video codec and SDP codec settings.
2N Indoor Units – general parameters and count of identified LAN devices.

SIP 1 and SIP 2

2N^® Clip allows two SIP accounts to be configured.
Allow SIP account – allow the SIP account to be used for calling. If disallowed, the account cannot be used for making outgoing calls and receiving incoming calls.

Device Identity

Display Name – set the name to be displayed as CLIP on the called party’s phone.
Phone Number (ID) – set the device phone number (or another unique ID composed of characters and digits). Together with the domain, this number uniquely identifies the device 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 dialog box enabling you to make a test call to a selected phone number, see below.

Authentication

Use Authentication ID – select the use of an alternative ID for device authentication. Otherwise, the Phone number value defined above is used.
ID for authentication – select the use of an alternative user ID for device authentication.
Password – set the password for device authentication. If your PBX requires no authentication, the parameter will not be applied.

SIP Proxy

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

SIP Registrar

Allow Registration – allow the device to be registered 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. The address is used where the main Registrar fails to respond to requests.
Backup Registrar Port – set the backup SIP Registrar port (typically 5060).
Registration Expiry – set the registration expiry, which affects the network and SIP Registrar loads by periodically sent registration requirements. The SIP Registrar can modify the expiry limit without letting you know.
Registration Status – display the current registration status (Unregistered, Registering..., Registered, Registration completing...).
Cause of Failure – display the cause of the last registration attempt failure – the Registrar’s last error response, e.g. 404 Not Found.

Advanced Settings

SIP Transport Protocol – set the SIP communication protocol: Choose UDP (default), TCP or TLS.
Lowest Allowed TLS Version – set the lowest TLS version to be accepted for server registration and connection setup.
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 Port for SIP – set the local port to be used for SIP signaling. A change of this parameter will not be applied until the device is restarted. The default value is 5060.
PRACK Allowed – enable the PRACK method (reliable confirmation of SIP messages with codes 101–199).
REFER Allowed – enable the REFER method for call redirection.
Send Keep Alive Packets – define whether or not the device shall periodically inquire about the called station status via SIP OPTIONS requests during calls (used for detection of the station failure during calls).
Allow IP Address Filter – enable the blocking of SIP packet receiving from addresses other than the SIP Proxy and SIP Registrar. The primary purpose of the function is to enhance communication security and eliminate unauthorized phone calls.
Accept Encrypted Calls Only (SRTP) – set that the 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 the MKI (Master Key Identifier), which is required by the counterparty for primary key identification if multiple keys are rotated in SRTP packets.
Do Not Play Incoming Early Media – disable playing of the incoming video stream before the call is picked up (early media), which is sent by some PBXs or other devices. A standard local ringtone is played instead.
QoS DSCP Value – set the priority of SIP packets 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 device is 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 SIP 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 PBX Compatibility – 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.
SRV Record Rotation – 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

Call Time Limit – set the call time limit after which the call is automatically terminated. The device beeps 10 s before the call ends to signal that the call end is approaching. If the call duration is set to 0 and SRTP is not used, the call is not time limited.

Incoming Calls

Call Receiving Mode (SIP 1/2, SIP 2) – set the way of receiving incoming calls. You can set the call receiving mode for each SIP account separately. The following three options are available:

- Always Busy – the device rejects incoming calls.
- Manual Answering – the device rings to signal incoming calls and the user can press a button to pick up.
- Automatic – the device picks up incoming calls automatically.

Local Call Receiving Mode – set the way of receiving incoming local calls.

- Always Busy – the device rejects incoming calls.
- Manual Answering – the device rings to signal incoming calls and the user can press a button to pick up.
- Automatic – the device picks up incoming calls automatically.

Do Not Disturb Mode with Timeout – ringing is disabled temporarily. If this silent mode is activated, the device shall not ring when a call comes in.

Outgoing Calls

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 longer than 20 s.
Ring 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

Audio Codecs

Enable/disable the use of audio codecs for call setups and set their priorities in this block.

DTMF Sending

This block helps you define how DTMF characters shall be sent from the device. Check the opponent’s DTMF receiving options and settings to make the function work properly.
In-Band (Audio) – enable classic DTMF dual tone sending in the audio band.
RTP (RFC-2833) – enable DTMF sending via the RTP according to RFC-2833.
SIP INFO (RFC-2976) – enable DTMF sending via SIP INFO messages according to RFC-2976.

DTMF Receiving

This block helps you define how DTMF characters shall be received from the intercom. Check the opponent’s DTMF receiving options and settings 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.

Transmission Quality Settings

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.

Video

Video Codecs

Enable/disable the use of video codecs for call setups and set their priorities.

Extended Codec Settings

Enabled – enable the packetization mode and set the payload type for each codec. The payload type will be selected automatically in case it cannot be set manually.
SDP Payload Type – set the "payload type" for video codec H.264 (packetization mode 1). Set a value from the range of 96 through 127, or 0 to disable this codec type.

Local Calls

Enable Local Calls – enable calls between 2N devices in the LAN. With this function off, the other LAN devices cannot locate this device, i.e. cannot call the device in the device:device_ID format.

Network Identification

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.

Connection to Intercoms

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 the intercoms fail to match, the devices cannot communicate, i.e. the intercom cannot call the 2N answering unit and vice versa.

Connection to Answering Units

Access Key – set the access key shared by the 2N answering units and intercoms. If the defined key in the answering units and the intercoms fails to match, the devices cannot communicate, i.e. the intercom cannot call the answering unit and vice versa.
Multicast address – set the multicast address to which messages shall be sent between the answering units.

LAN Devices

LAN Device Count – display the count of devices in the LAN.
Show list of LAN devices – show a detailed list of LAN devices.