Integration of Open Source and Enterprise IP PBXs

Over the last few years, voice over Internet protocol (VoIP) evolved from enabling voice communications between computer terminals to providing much wider functionality of the public switched telephony network (PSTN). The idea of deploying VoIP in private branch exchanges (PBXs) is gaining credence among service providers, and this paper addresses some of the design issues related to an IP PBX implementation. Specifically, the paper explores the interoperability between Cisco call manager (CCM) and Linux-based asterisk PBXs corresponding to the enterprise-class and open source IP PBXs, respectively. The architecture of the networking testbed is documented for running various VoIP protocols between multi-vendor devices and IP PBXs. The results presented can be broadly defined into three areas of investigation: (i) the use of session initiation protocol (SIP) and inter asterisk exchange (IAX) trunks to integrate CCM to Asterisk PBXs and Asterisk to Asterisk PBXs, respectively; (ii) examining interoperability between the CCM and Asterisk PBX with advanced features like call display, voicemail, call detail records (CDR) and voicemail to e-mail transfer; and (iii) testing the resilience of voice quality against packet loss and delay in the open source PBX trunks with National Institute of Standards and Technology (NIST) network simulator.