FreeSWITCH is gearing up for the 10th annual ClueCon this August. The guys wanted to take a chance to stop and hang out with the VUC and give a preview of what’s on tap for this summer. FreeSWITCH 1.4 has been released with a whole bunch of new security features including SIP over secure WebSockets and GCM mode SRTP. The new version also includes a fully functional WebRTC media engine. This week we will also announce 2 new big WebRTC related projects being worked on inside FreeSWITCH.
FreeSWITCH was originally designed and implemented by Anthony Minessale with the help of Brian West and Michael Jerris. All 3 are former developers of the popular Asterisk open source PBX. The project was initiated to focus on several design goals including modularity, cross-platform support, scalability and stability. Today, many more developers and users contribute to the project on a daily basis.
FreeSWITCH supports many advanced SIP features such as presence/BLF/SLA as well as TCP TLS and sRTP. It also can be used as a transparent proxy with and without media in the path to act as a SBC (session border controller) and proxy T.38 and other end to end protocols.
FreeSWITCH supports both wide and narrow band codecs making it an ideal solution to bridge legacy devices to the future. The voice channels and the conference bridge module all can operate at 8, 12, 16, 24, 32 or 48 kilohertz and can bridge channels of different rates. The G.729 codec is also available under a commercial license.
FreeSWITCH builds natively and runs standalone on several operating systems including Windows, Max OS X, Linux, BSD and Solaris on both 32 and 64 bit platforms.