Understanding SIP call routing

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When the PortaSIP server has to establish an outgoing call, it must find out where the call is being sent to. To do this, it will ask billing for a list of possible routes. In this case the routing configuration is in one central location, and billing can use information about termination costs, quality or other parameters to choose the best route (least-cost routing, quality-based routing, profit-guarantee, individual routing plans, etc.).

When a call goes through the PortaSIP server, the SIP server may:

  • Direct the call to one of the registered SIP clients, if the called number belongs to the registered agent.
  • Optionally, direct the call to the voicemail box (the Media Server required) if the called number belongs to an account in PortaBilling, but this account is not currently registered to the SIP server (is offline).
  • Route the call to one of the gateways for termination, according to the routing rules specified in PortaBilling.

Please consult PortaBilling Administrator Guide for more information about various routing parameters and methods.

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