3.2.3.2 SIP 1 / SIP 2

Two SIP accounts can be configured on 2N device  . Thus, the device can be registered under two phone numbers, with two different SIP exchanges and so on. Both the SIP accounts process incoming calls equivalently. Outgoing calls are primarily processed by account SIP 1, or, if account SIP 1 is not registered (due to SIP exchange error, e.g.), by account SIP 2. Select the account number for the phone numbers included in the phone directory to specify the account to be used for outgoing calls (example: 2568/1 - calls to number 2568 go via account SIP 1, sip:1234@192.168.1.1 calls to sip uri via account SIP 2).

Configuration

  • 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 as CLIP on the called party's phone.
  • 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 device in calls and registration.
  • Domain – set the domain name of the service with which the device is registered. Typically, it is equivalent to the SIP Proxy or Registrar address.
  • Test Call – display a dialogue window enabling you to make a test call to a 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 device 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. The address is used where 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. 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 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, Unregistering...).
  • Failure Reason – display the reason for the last registration attempt failure: the registrar’s last error reply, 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. 
  • STUN Enable – enable STUN functionality for the SIP account. Address and ports acquired from the configured STUN server will be used in SIP headers and SDP media negotiation.
  • STUN Server Address – set the IP address of the STUN server that will be used for this SIP account.
  • STUN Server Port – set the port of the STUN server that will be used for this SIP account.
  • 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.
  • Compatibility With Broadsoft Devices – 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.

Video


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


  • H.264 Baseline Profile, Packetization Mode 1
  • H.264 Baseline Profile, Packetization Mode 0
  • H.264 Main Profile, Packetization Mode 1
  • H.264 Main Profile, Packetization Mode 0
  • H.264 High Profile, Packetization Mode 1
  • H.264 High Profile, Packetization Mode 0
  • H.264 Constrained Baseline Profile, Packetization Mode 1
  • H.264 Constrained Baseline Profile, Packetization Mode 0
    • Enabled – enable the packetization mode and set the payload type for each codec. The payload type can 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 option.

Audio

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

The tab 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 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.


The tab 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 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. A higher capacity improves the transmission resistance at the cost of a greater sound delay.