4.3 Calling

Calling is the basic function of 2N® LiftIP 2.0 allowing you to establish connections with other IP network terminals. The device supports the extended SIP and is compatible with major SIP PBX and terminal equipment manufacturers. 

2N® LiftIP 2.0 uses the following protocols for audio stream encoding (or compression): G.711, L16, G.722 and G.729. 

Explanation of IP Telephony Terms

  • SIP (Session Initiation Protocol) – a phone call signaling transmission protocol used in IP telephony. It is primarily used for setting up, terminating and forwarding calls between two SIP devices (the device 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 – 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 – 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) – a protocol defining the standard packet format for audio/video transmission via IP networks. The device uses this protocol for audio and video stream transmissions during a call. The stream parameters (port numbers, protocols and codecs) are defined and negotiated via the SDP (Session Description Protocol).

2N® LiftIP 2.0 supports three SIP signaling types:

  • Using the UDP, which is the most common unsecured way of signaling,
  • Using the TCP, which is a less widely used, yet recommended way of unsecured signaling;
  • Using the TLS protocol, where SIP messages are secured against eavesdropping and modifications by third parties.

Here is what you can find in this section: