These features provide rerouting of SIP calls if a gateway cannot accept an outbound call (connection request). The calls can be off-net (to the PSTN via a SIP gateway) or on-net (to another SIP network, such as Telephony Application Server). The SIP response codes upon which rerouting is attempted are provisioned in the system, to provide flexibility when dealing with various third-party gateway and server types.
In addition, a rerouting timer provides rerouting to handle the case when no response is received from the remote SIP gateway or SIP server after an INVITE has been sent.
These features can also be configured to provide for the rerouting of SIP calls between SIP subscribers through the PSTN, in case of WAN failure between the two subscribers, or after receipt of a SIP response code indicating bandwidth congestion on the WAN link.
These features monitor the gateway so that a more intelligent routing can take place. Subsequent calls are immediately sent to the next available route, and polling begins on the unreachable gateway to determine when the problem or congestion is resolved. Call routing automatically switches back to the gateway when the polling mechanism indicates the problem or congestion is resolved.
Gateway rerouting may be triggered when a call destined for a gateway or peer server - for example, another softswitch in the network - cannot be completed because this destination is unreachable as a result of a malfunction, congestion, or LAN/WAN link outage.Parent topic: CAC (Call Admission Control) Rerouting
Subscriber rerouting may be triggered when a call destined for a remote subscriber - for example, in a branch office - is blocked by congestion or outage of the LAN/WAN link. When subscriber rerouting occurs, it may or may not lead to gateway rerouting. Beginning in legacy system, subscriber rerouting accommodates the needs of customers that need access to certain Lotus Sametime Unified Telephony features - for example, group features that are needed by subscribers in a branch.