5.4.1 Phone


The Phone service is one of the basic functions of the intercom: helps you establish connections with other IP network terminal equipment. The 2N IP intercoms support the extended SIP and are compatible with and certified by the leading SIP PBX and terminal equipment manufacturers (CISCO, Avaya, Broadsoft, etc.).

The intercom supports up to five parallel calls: 1 outgoing and up to 4 incoming calls. Just one of the calls can be active – the audio stream is interconnected with the microphone and speaker and video stream with the camera. The other calls are always inactive – the microphone and speaker are muted, the intercom receives the DTMF characters for the opponent to control the intercom (activate/deactivate profiles, users, etc.).

Typically, the intercoms are used for outgoing calls and incoming calls are inactive – the microphone and speaker are muted. However, you can configure your intercom to make incoming calls active and ringing; refer to the Calls tab. Press the * and # keys on the numeric keypad to answer and terminate an incoming call. 

The 2N IP intercoms use the G.711L16, G.722 and G.729 protocols to encrypt or compress audio streams and the H.263 or H.264 codecs to compress video streams. Broadband codecs L16 and G.722 are available in selected 2N IP intercom models only. Choose your preferential codecs in the Audio or Video tab.


Explanation of IP Telephony Terms

  • SIP (Session Initiation Protocol) – is a phone call signalling transmission protocol used in IP telephony. It is primarily used for setting up, terminating and forwarding calls between two SIP devices (the intercom and another IP phone in this case). SIP devices can establish connections directly with each other (Direct SIP Call) or, typically, via one or more servers: SIP Proxy and SIP Registrar.
  • SIP Proxy – is an IP network server responsible for call routing (call transfer to another entity closer to the destination). There can be one or more SIP Proxy units between the users.
  • SIP Registrar – is an IP network server responsible for user registration in a certain network section. As a rule, SIP device registration is necessary for a user to be accessible to the others on a certain phone number. SIP Registrar and SIP Proxy are often installed on one and the same server.
  • RTP (Real-Time Transport Protocol) – is a protocol defining the standard packet format for audio and video transmission in IP networks. 2N IP intercom uses the RTP for audio and video stream transmission during a call. The stream parameters (port numbers, protocols and codecs) are defined and negotiated via the SDP (Session Description Protocol).

The 2N IP intercoms support three ways of SIP signalling:

  • via the User Datagram Protocol (UDP), which is the most frequently used unsecured signalling method
  • via the Transmission Control Protocol (TCP), which is less frequent, yet recommended unsecured signalling method
  • via the Transaction Layer Security (TLS) protocol, where SIP messages are secured against third party monitoring and modification (except models 2N® IP Base, Uni)

List of Parameters

The 2N IP intercom Phone settings are arranged in five tabs:

  • SIP 1 and SIP 2 – complete SIP terminal settings
  • Calls – incoming and outgoing call settings
  • Audio – audio codec, DTMF transmission and other audio stream transmission settings
  • Video – video codec, video resolution and other video stream transmission settings
  • Local Calls – set the local calls including connections, video parameters 
  • Calling to ACS – set the Axis Camera Station calls

SIP 1 and SIP 2

The 2N IP intercoms allow two independent SIP accounts (SIP 1 and SIP 2 tabs) to be configured. Thus, the intercom 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 1, or, if account 1 is not registered (due to SIP exchange error, e.g.), by account 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 1, sip:1234@192.168.1.1 calls to sip uri via account 2).


  • 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 the intercom phone number (or another unique ID including characters and digits). Together with the domain, this number represents a unique intercom identification in calls and registration. 
  • Domain – set the domain name of the service with which the intercom is registered. Typically, it is identical with 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. 




  • Use Authentication ID – enable the use of an alternative ID for intercom authentication. If disabled, the phone number defined above is used for authentication.
  • Authentication ID – enter the alternative ID for authentication.
  • Password – enter the password for authentication. The parameter is applied on if your PBX requires authentication.



  • Proxy Address – set the SIP Proxy IP address or domain name.
  • Proxy Port* – set the SIP Proxy port. The device uses the default port according to the transport layer (5060 or 5061) or a port obtained from DNS in case the parameter is empty or set to 0.
  • Backup Proxy Address – set the SIP Proxy IP address or domain name to be used where the main proxy fails to respond to requests.
  • Backup Proxy Port* – set the backup SIP Proxy port. The device uses the default port according to the transport layer (5060 or 5061) or a port obtained from DNS in case the parameter is empty or set to 0.



  • Registration Enabled – enable intercom registration with the set SIP Registrar.
  • Registrar Address – set the SIP Registrar IP address or domain name.
  • Registrar Port* – set the SIP Registrar port. The device uses the default port according to the transport layer (5060 or 5061) or a port obtained from DNS in case the parameter is empty or set to 0.
  • Backup Registrar Address – set the SIP registrar IP address or domain name to be used where the main registrar fails to respond to requests.
  • Backup Registrar Port* – set the backup SIP registrar port. The device uses the default port according to the transport layer (5060 or 5061) or a port obtained from DNS in case the parameter is empty or set to 0.
  • Registration Expires – define the registration expiry, which affects the network and SIP Registrar load by periodically sent registration requirements. The SIP Registrar can modify the expiry limit 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 last error reply of the registrar, e.g. 404 Not Found.
     

Tip

  •  To set the Outbound Proxy complete the Outbound Proxy address into the Proxy address and Registrar address parameters. Domain = Registrar address.

Caution

  • If the parameter* is empty or set to 0, the default port is used according to the selected transport protocol (5060 for TCP or UDP, 5061 for TLS).

  • SIP Transport Protocol – set the SIP communication protocol: UDP (default), TCP or TLS.
  • Lowest Allowed TLS Version – define the lowest TLS version to be connected to the devices.
  • 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 to be used for SIP signalling. The parameter is not applied until the intercom is 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 KeepAlive Packets – define whether the intercom shall, during a call, send periodical SIP OPTIONS requests to inquire about the state of the called station (to detect the station failure, e.g.).
  • 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 unauthorised 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 call pick-up (early media), which is sent by some PBXs or other devices. 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. Value is entered in decimal format. The parameter is not applied until the intercom is restarted.
  • Starting RTP Port – set the starting local RTP port in the range of the length of 64 ports to be used for audio and video transmissions. The default value is 4900 (i.e. the used range is 4900–4963). The parameter is only set for account 1 but applies to both the SIP accounts.
  • External IP Address – set the public IP address or name of the router to which your intercom is connected. If the intercom IP address is public, leave this field blank.
  • 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 is terminated by the intercom. Set the parameter to 0 to disable this function. The parameter is only set for account 1 but applies to both the SIP accounts.
  • 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 record – 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.

Caution

  • To use the NAPTR / SRV DNS query, cancel the Proxy/Registrar port setting.

Calls



  • Call Time Limit – set the call time limit after which the call is automatically terminated. The intercom signals termination with a 10s beep before the call end. Enter any DTMF character into the call (# on your IP phone, e.g.) to extend the call time. If the call duration is set to 0 and SRTP is not used, the call is not time limited.



  • Call Answering Mode(SIP1, SIP2) – set the incoming call receiving mode. The following three options are available:
    • Always busy – the intercom rejects incoming calls,
    • Manual – the intercom alerts incoming calls and the user answers them using a numeric keypad button, and 
    • Automatic – the intercom answers incoming calls automatically. You can set the call receiving mode for each SIP account separately.
    • Automatic (DTMF only) – the intercom answers incoming calls automatically only if DTMF without connection to a microphone and speaker is received.
  • Local Call Receiving Mode – set the incoming local call receiving mode
    • Always busy – the intercom rejects incoming calls,
    • Manual – the intercom alerts incoming calls and the user answers them using a numeric keypad button, and 
    • Automatic – the intercom answers incoming calls automatically. You can set the call receiving mode for each SIP account separately.
    • Automatic (DTMF only) – the intercom answers incoming calls automatically only if DTMF without connection to a microphone and speaker is received.
  • Pick Up in – set the timeout after which the call is automatically picked up in the automatic call answering mode. If one of the Answering machine modes is enabled in an Answering machine supporting device, the call is picked up after the timeout and the selected voice message is played in both the automatic and manual call answering modes. If this value is 0, the voice message is played instantaneously. Shared by all the SIP accounts.
  • Answer Incoming Call by Button – pick up an incoming call via a selected speed dial button. Set to None to disable the function.

Caution

  • The Answer Incoming Call by Button function is not displayed in the keypad-equipped 2N® IP Force and 2N® IP Vario models. With these models, answer incoming calls by pressing the green-earpiece button on the keypad without prior configuration.


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.

Ring Time Limit – set the outgoing call setup and ringing time limit after which the calls shall be 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. Minimum value 1 s, maximum value 600 s. Configure 0 to disable this time limit.Dial Cycles Limit – set the maximum count of user deputy dial cycles if the user dialled by the Phone Book position number is inaccessible. The function helps you avoid deadlock if the User Deputy is set to the same value in the Phone Book. Refer to Subs. 5.4.1.1 Calling Cycle Limit for calling cycle limit settins options.Calling Virtual Numbers – allow the calling of preset virtual numbers of users.Floor/Apartment Dialing Mode

 enable the special Floor/Apartment dialling mode. In this mode, enter the assigned user virtual number via the numeric keypad. Available for model 2N® IP Vario only. Enter the floor/apartment code to the user Virtual number. The code may include digits and letters A–F.Telephone Mode Enabled  enable the option to set up calls directly to the phone numbers dialed via the intercom numeric keypad. Enter the phone number key sequence to set up the call.


Tip

  • Set up a call to 2N® IP Force and 2N® IP Vario as follows: press phone_number  (or  phone_number  for 2N® IP Verso). If you do not press  (or for 2N® IP Verso) as the terminating character, the dialling will be confirmed automatically when the code entering timeout expires as if  (or  for 2N® IP Verso) was pressed.
  • Maximum Number of Dialed Digits  Set the maximum count of digits for a phone number in the Telephone mode. When this limit is reached, the number is dialed automatically without pressing *.
  • Button Function During Outgoing Call – set the quick dial button function during an outgoing call. You can only set the button that initiated the call.



  • Enable Crestron Network Discovery – enable 2N IP intercom identification within the Crestron network.
  • Crestron Device Name – select the device name.
  • Crestron Group List – select the group name list with comma as a separator.
  • Enable Video Multicast for Crestron panels – enable video multicast for Crestron panels, allowing for multiple Crestron devices to receive the same video stream without wasting the local network bandwidth.
  • Crestron Multicast Address – set the multicast address to be used for multicast video for Crestron devices.
  • Crestron Multicast Port – set the multicast port to be used for multicast video for Crestron devices.
  • Crestron Multicast TTL – set the Time To Live (TTL) value to be used for sending video early media for Crestron devices. 

Audio



  • Enable/disable the use of audio codecs for call setups and set their priorities. Broadband codecs L16 and G.722 are available in selected intercom models only. Codec G.729 is available for all the 2N IP intercoms.

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


  • Sending Mode – define whether it will be possible to send DTMF during a call by pressing 0 through 9, * and # on the intercom numeric keypad. Set the sending mode for incoming/outgoing/all calls.
  • 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 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. Value is entered in decimal format. The parameter is not applied until the intercom is restarted.
  • Jitter Compensation – set the buffer capacity for jitter compensation in audio packet transmissions. A higher capacity improves the transmission resistance at the cost of a greater sound delay.

Video



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



  • Video Resolution – set the video resolution for phone calls.
  • Video Framerate – set the video frame rate for phone calls.
  • Video Bitrate – set the video stream bit rate for phone calls.



  • PTZ Mode – enable the PTZ (Pan-Tilt-Zoom) function to control the camera display area during the call via DTMF (GOLD license required) from your IP phone numeric keypad. Click the * key to enable/disable the PTZ mode. The meanings of the IP phone keys in the PTZ mode are as follows:

    IP phone key
    PTZ mode function
    *Enable/disable PTZ
    1Zoom in
    3Zoom out
    2Move zoom region up
    4Move zoom region to the left
    6Move zoom region to the right
    8Move zoom region down
    5Return to initial state



  • QoS DSCP Value – set the video RTP packet priority in the network. The set value is sent in the TOS (Type of Service) field in the IP packet header.
  • Maximum Packet Size – set the size limit for the video RTP packets to be sent.



  • H.264 Payload Type (1) – set the payload type for video codec H.264 (packetisation mode 1). Set a value from the range of 96 through 127, or 0 to disable this codec type.
  • H.264 Payload Type (2) – set the payload type for video codec H.264 (packetisation mode 2). Set a value from the range of 96 through 127, or 0 to disable this codec type.
  • H.263+ Payload Type – set the payload type for video codec H.263+ (packetisation mode 3). Set a value from the range of 96 through 127.
  • Use the sendrecv attribute for video – the setting was earlier named Compatibility with Polycom phones. This setting provides compatibility with some third party devices (Polycom/Cisco and others). In this mode, the intercom sends sendrecv instead of sendonly in the SDP message in the codec offer for video.

Tip

  • For the Video Preview feature at the Grandstream GXV 3275 phone (video transferred via Early Media) no configuration is needed. Check your PBX vendor whether this feature is supported by your PBX system.
  • For the Video Preview feature at the Gigaset Maxwell 10 phone (video transferred via jpg images) it is necessary to set Connection Type to Unsecure and Authentication to None at the Camera API in HTTP API.


  • Enable Incoming Video – if this mode is on, the intercom displays the opponent’s video during a call if the other party allows so.
  • Incoming Video Aspect Ratio – set the preferred incoming video aspect ratio to be displayed. If an aspect ratio other than the default one is selected, the video is cropped to fill the whole display width in the new aspect ratio.
  • Display Outgoing Video – select whether or not the intercom shall display the preview of the video to be sent during a call.

Local Calls

This tab contains settings for connection of the 2N answering units to the intercom. The main parameter is the access key, which secures the connection and enables you to create multiple independent groups of intercoms and 2N answering units within the local network. It also contains the video transmission settings.


  • 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.


  • 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 1–3 – set the access key to be shared by the intercom and 2N answering unit. If the access keys do not match in the intercom and2N answering unit, the intercom cannot call the 2N answering unit and the 2N answering unit cannot receive video from the intercom. Each intercom can be assigned up to three access keys and thus become a member of up to three independent 2N answering unit groups. The Access key length is up to 63 characters. 


  • The access key cannot be used with 2N® Indoor Touch firmware v. 2 or 3 where it has to be set as empty. The access key can only be used for 2N® Indoor Touch version 4 or higher.

 

  • Video Resolution – set the resolution of the video stream to be sent to 2N answering unit.
  • Video Framerate – set the framerate of the video stream to be sent to 2N answering unit.
  • Video Quality – set the quality of the MJPEG video stream to be sent to 2N answering units.
  • Multicast Group – set the multicast address to which the intercom video stream shall be sent. Select one of the 8 preset addresses or set the mode in which the intercom selects the address automatically. 
  • Enable Video Preview – enable video preview multicast transmission.


  • LAN Device Count– display the current count of local 2N answering units connected to the intercom, i.e. those registered with the intercom.
  • Number of Listening/Watching Devices – display the current count of 2N answering units watching video streams from the intercom.
  • Show LAN device list  display the list of local 2N answering units.



ACS

This function is used for integration of the Axis Camera Station service into the 2N IP intercoms.


  • Enable ACS Call – allow calling to the Axis Camera Station (ACS). Use a special URI in the vms:* format for the ACS calls.

Caution

  • In case the 2N IP intercom has already been added to the ACS, back up all of its records before upgrade, then remove the 2N IP intercom from the ACS, perform upgrade and add the intercom again.



  • Username – ACS call authentication username.
  • Password – ACS call authentication password.