Session Initiation Protocol (SIP) is a signaling protocol, which is widely used for setting up, connecting and disconnecting communication sessions, mostly voice or video calls over the Internet (VoLTE). SIP is a standardized protocol with its basis coming from the IP community and in most cases uses SIP is becoming an industry standard in that although there are other signalling protocols, SIP is now the dominate one. However, here is the SIP trunking tutorial you which going to help you out.
SIP Trunking Tutorial: Deploy
The SIP trunk can connect your organization to an ‘ITSP’ through an IP-PBX or a Mediation Server. A Mediation Server performs encryption, decryption, and data translation between an Office Communications.
Every Mediation Server has two interfaces (network adapters): an internal interface & an external interface. The external interface is generally called as the gateway interface as until recently it was usually connected to an IP-PSTN gateway (or the IP-PBX). When you deploy SIP trunking, the external interface is connected to the SIP trunk.
SIP Trunking Tutorial: Applications,
SIP is an application-layer control protocol that supports five parts of stopping and making communications. It doesn’t provide services. Therefore it acts with the other protocols to provide these services, one of which is typically RTP that carries a voice for a call.
The five parts of setting up and terminating calls that SIP Trunking handles are:
- User Location: It determines where the end system is that will be used for a call.
- User Availability: It determination of the willingness (availability) of the called party to engage in a call.
- User Capabilities: It determination of the media and parameters which will be used for the call.
- Session Setup: The establishment of the session parameters from both parties (ringing).
- Session Management: Invoking the services including transfer, termination, and modifying the sessions settings.