SIP trunking


SIP trunking enables the end point’s PBX to send and receive calls via an IP network, such as the Internet or private WAN. As SIP is applied for the signalling protocol for multiple real-time application, SIP trunk is able to control voice, video and messaging applications. It is also a voice over Internet Protocol technology and streaming media service based on the Session Initiation Protocol by which Internet telephony service providers deliver telephone services and unified communications to customers equipped with SIP-based IP PBX and unified communications facilities. Most unified communications applications provide voice, video, and other streaming media applications such as desktop sharing, web conferencing, and shared whiteboard.

Domains

The architecture of SIP trunking provides a partitioning of the unified communications network into two different domains of expertise:
The interconnection between the two domains must occur through a SIP trunk. The interconnection between the two domains, created by transport via the Internet Protocol, involves setting specific rules and regulations as well as the ability to handle some services and protocols that fall under the name of SIP trunking.
The ITSP is responsible to the applicable regulatory authority regarding all the following law obligations of the public domain:
The private domain instead, by nature, is not subject to particular constraints of law, and may be either the responsibility of the ITSP, the end user, or of a third party who provides the voice services to the company

Architecture

Each domain has elements that perform the characteristic features requested of that domain, in particular the result is logically divided into two levels:
The private domain consists of three levels: