With increasing digitization and proliferation of IP networks, use of SIP protocol to provide IP telephony services has seen a steady rise. In IP telephony networks, call continuity in the event of a failover is of primary importance.
The paper presents design improvement proposal on top of the patented “Failover Management of SIP-based Multimedia Communication Sessions” [1] work to increase the call-continuity ratio and reduce the time a secondary server currently takes to take over the call for better user experience, setup and call flows with media(RTP) passing through the SIP server.
Introduction
With the current patented [1] solution design, we are able to provide call concurrency in failover cases for SIP-based multimedia communication sessions.
The patented[1] solution design provides an approach for call continuity in a deployment where SIP server acts as a Signaling Gateway Application. This may not require endpoints to be REGISTERED and media to be traversed via the SIP server, as may be required in other deployments such as Softswitch, (Hosted) IPPBX.
The patented design also does not take into account the need for starting secondary server’s service beforehand i.e. before the handover takes place in the event of a failover. It takes considerable time for the secondary server to initiate services and this delays the call continuity handover.
