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SIP Trunking Print E-mail

 

SIP (Session Initiation Protocol) Trunking is replacing ISDN connections for premise-based telephony systems (PBXs).  Calls are sent and received over an IP connection, which is often a broadband (ADSL) circuit.  The definition of a SIP Trunk is a single concurrent call over an IP network.  Two concurrent calls would require two SIP Trunks.

 

Business benefits

Principally cost savings, plus the ability to implement a converged IP network and take advantage of Unified Communications benefits, such as Presence.

 

  • Free VoIP calls between sites on the Private Internet
  • On-demand capacity to meet demand
  • Hardware savings compared with analogue or ISDN ports
  • Centralised PSTN access across the Enterprise
  • Compatible with local ISDN for added resilience
  • Enterprise-wide billing, including home-workers
  • Advantageous call tariffs

 

Session Initiation Protocol (SIP) enables a call to be set-up on an IP network between two or more applications.  The call can be any type of real-time information; data, voice (VoIP) or multi-media.

 

The bandwidth of the IP link(s) to the PBX is sized to handle the maximum number of concurrent SIP Trunks (calls).  Additional SIP Trunks can be provisioned on-demand, so long as there is enough bandwidth available on the link.

 

Organisations with multiple sites typically deploy a private IP network.  Each PBX needs only a single LAN port, because the IP network provides the inter-PBX routing, making the IP solution much cheaper than traditional multiple trunk analogue or ISDN configurations.

 

Local IP gateways enabled non-IP PBXs to connect to a SIP Trunking service, although local ISDN circuits can still be supported to support the migration, or for added service resilience if required.

 

Calls connect to the public switched telephone network (PSTN) via a gateway, which is provided by the SIP Trunk service provider within the core of the IP network.  This results in simplified billing, especially for home-workers.

 

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