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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 Indoor Talk 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 example below: the 2N® Indoor Talk is registered to the SIP Proxy (2N Netstar SW) 192.168.50.252, The phone number ID is 200, SIP Proxy and SIP proxy registrar Server is also 192.168.50.252 and there is used a default port 5060. The Authentication ID and password is in case also required from the Pbx.

It is possible to check the current registration status in the Section Status-Services - Registration state-REGISTERED

We can now define a call destination in the section Hardware-Buttons and assign User Phone number regarding to dialing plan of the SIP Proxy server.