Both the SIP accounts process incoming calls equivalently. Thus, the intercom can be registered under two phone numbers, with two different SIP exchanges and so on. The 2N IP intercoms allow two independent SIP accounts (SIP 1 and SIP 2 tabs) to be configured.
Calling to ACS – set the Axis Camera Station calls. Local Calls – set the local calls including connections, video parameters. Video – video codec, video resolution and other video stream transmission settings. 
Audio – audio codec, DTMF transmission and other audio stream transmission settings.Calls – incoming and outgoing call settings.SIP 1 and SIP 2 – complete SIP terminal settings.The 2N IP intercom Phone settings are arranged in five tabs: via the Transaction Layer Security (TLS) protocol, where SIP messages are secured against third party monitoring and modification (except models 2N ® IP Base, Uni).
via the Transmission Control Protocol ( TCP), which is less frequent, yet recommended unsecured signalling method. 
via the User Datagram Protocol ( UDP), which is the most frequently used unsecured signalling method. The 2N IP intercoms support three ways of SIP signalling: The stream parameters (port numbers, protocols and codecs) are defined and negotiated via the SDP (Session Description Protocol). 2N IP intercom uses the RTP for audio and video stream transmission during a call. RTP (Real-Time Transport Protocol) – is a protocol defining the standard packet format for audio and video transmission in IP networks. SIP Registrar and SIP Proxy are often installed on one and the same server. As a rule, SIP device registration is necessary for a user to be accessible to the others on a certain phone number.
SIP Registrar – is an IP network server responsible for user registration in a certain network section. There can be one or more SIP Proxy units between the users. SIP Proxy – is an IP network server responsible for call routing (call transfer to another entity closer to the destination). SIP devices can establish connections directly with each other (Direct SIP Call) or, typically, via one or more servers: SIP Proxy and SIP Registrar. 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 (Session Initiation Protocol) – is a phone call signalling transmission protocol used in IP telephony. 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. The 2N IP intercoms use the G.711, L16, G.722and G.729 protocols to encrypt or compress audio streams and the H.263 or H.264 codecs to compress video streams. Press the * and # keys on the numeric keypad to answer and terminate an incoming call. However, you can configure your intercom to make incoming calls active and ringing refer to the Calls tab. Typically, the intercoms are used for outgoing calls and incoming calls are inactive – the microphone and speaker are muted. 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.). 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 intercom supports up to five parallel calls: 1 outgoing and up to 4 incoming calls. 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 Phone service is one of the basic functions of the intercom: helps you establish connections with other IP network terminal equipment.
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