IP aliasing in PortaSIP

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IP aliasing is a simplest way to migrate customers from external providers to PortaSIP.

Consider the following example. A service provider uses PortaSIP in their network for voice calls services. This service provider then acquires some new customers but their IP phones are configured to use another provider’s SIP server – and the service provider would like to migrate them to PortaSIP.

With IP aliasing functionality, the tedious reconfiguration of each customer’s IP phone is not required. Instead, an administrator configures the IP address of the external SIP server as an alias to PortaSIP’s virtual IP address.

Along with the IP alias, the administrator can also define additional transport ports for the following protocols: UDP, TCP and TLS. This step is optional and only required if standard transport ports for these protocols are blocked for some reason or cannot be used.

The IP aliasing functionality significantly simplifies the migration procedure to PortaSIP and makes the entire process of migration fully transparent for end users.

Call delivery to IP phones registered via IP alias
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  • User A’s SIP phone registers with PortaSIP via an IP alias and the following record is created in the registration database:
    sip:User A@IP alias:port
  • User B, whose SIP phone is registered to PortaSIP via the main (virtual) IP address, dials user A’s phone number and an INVITE request is sent with the following addresses-of-record (AORs) in the From: and To: headers:
    From:User B@IP address
    To:User A@IP address

    The INVITE request is delivered to the PortaSIP main IP address (1).

  • The dispatching node forwards the call to one of the available processing nodes (2).
  • The processing node sends an authorization request to PortaBilling (3) and receives an authorization response (4).
  • Once authorized successfully, the call is further processed. The B2BUA checks the registration database and detects that user A’s SIP phone is registered with an IP alias. The B2BUA therefore changes the AORs in the From: and To: headers as follows (5):
    From:User B@IP alias
    
    To:User A@IP alias
  • The B2BUA routes the call to the dispatching node (6).
  • Finally the dispatching node establishes the call with user A’s IP alias (7).

The key point in this scenario is that the same IP address and port for call delivery is used as what was used by PortaSIP during the recipient’s SIP phone registration.

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