FreeSwitch’s latest release
FreeSwitch, the open source VoIP platform designed for carriers, has announced the latest version. The biggest addition is a telephony abstraction library that normalizes several existing voice protocols into a single API. The main reason FreesSwitch developed module was to get H323 support. According to FreeSwitch, it is also possible to expand the module to support other protocols such as IAX2 and SIP.
The open source project has also added the Least Cost Routing module in its latest version. Some of the service providers using FreeSwitch include Teliax, tollfreegateway.com, Truphone, Paetec and a few more that are migrating to FreeSwitch but have not announced it yet.
There are three main open source VoIP platforms in use. These include Asterisk, FreeSwitch, and OpenSER. Asterisk is mostly in use in the enterprise environment. FreeSwitch and OpenSER are used by carriers. OpenSER is a proxy that routes SIP packets. FreeSwitch is a b2bua that carries media and interconnects networks. Both work together to form a complete system.
FreeSwitch developer community seems to be growing on a regular basis. It’s IRC channel (irc.freenode.net #freeswitch) has close to 200 members in it on a daily basis. There are over a thousand members on the mailing list. Both those numbers have doubled in the last 6 months.
