Open source VoIP scale up beautifully
Several smaller colleges in the U.S. and in Europe already have deployments in place based on Digium's Asterisk software private branch exchange (PBX) and on SIPfoundry IP-PBX systems. Several larger universities are collaborating on still more modular and scalable approaches in which the workload is partitioned, using Asterisk and/or SIPfoundry, but also using SER and OpenSER. Incidentally, SIP Express Router (SER) is included in many Linux distributions and in the Sun Solaris operating system. It also has been proven in numerous very large production deployments, including Freenet.de, the German telecom and ISP, which is serving more than 1 million endpoints, and FWD (formerly FreeWorld Dialup), with half a million endpoints. One development -- a collaboration among four East Coast U.S. universities -- has resulted in an approach and a collection of standard tools that could eventually be packaged as a basic "cookbook" to make deployments in other universities more straightforward. Meanwhile, the work on open standards for VoIP and supporting protocols continues at a strong clip, as does the work on key open source IP-telephony projects. Over time, it is very reasonable to expect more mature standards, more mature code, and as a result, more deployments based on open source and open standards. It's also likely that IP-telephony will start to migrate away from a predominantly handset-based service to a mixed mode of softphones and hardphones. The softphones will more easily enable integration with other real-time communications tools, and with enterprise presence services. This will in turn help to move telephony away from its traditional role, and toward increasing integration with other collaboration tools, until telephony becomes part of an enterprise's unified communications infrastructure. These information was gathered from an two part article published by TechNewsWorld.
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