Abstract: A technique for allowing a non-SIP user to call a SIP user includes dialing an established service number that indicates a desire to place a call to a SIP user. The SIP URI of the intended call recipient is included in a call setup protocol message associated with dialing the service number. A non-SIP network recognizes the call to the service number and the SIP URI from the UUI parameter of the call setup message. The call is then routed to a gateway for interfacing between the non-SIP network and the appropriate SIP network where the SIP URI is extracted from the message received by the gateway and used to generate an SIP INVITE message for establishing the call with the intended SIP user.

Abstract: In a seamless redundancy or failover system for an IP network, data intended for a master component is received at a seamless redundancy component, where the data is routed both to the master component and to a standby component. The standby component is configured to process the data in the same manner as the master component, e.g., the standby component may be a duplicate of the master component, or another component configured to perform the same data processing functions. For seamless redundancy/failover, the data output of the standby component is suppressed unless and until the master component enters a failure condition, at which time the data output of the standby component is enabled for transmission to a downstream network component. “Failure condition” refers to an operational state of the master component where the master component is unable to process received data in its intended and normal manner.

Abstract: A technique for allowing a non-SIP user to call a SIP user includes dialing an established service number that indicates a desire to place a call to a SIP user. The SIP URI of the intended call recipient is included in a call setup protocol message associated with dialing the service number. A non-SIP network recognizes the call to the service number and the SIP URI from the UUI parameter of the call setup message. The call is then routed to a gateway for interfacing between the non-SIP network and the appropriate SIP network where the SIP URI is extracted from the message received by the gateway and used to generate an SIP INVITE message for establishing the call with the intended SIP user.