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FreeSWITCH is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media. It was created in 2006 to fill the void left by proprietary commercial solutions. FreeSWITCH also provides a stable telephony platform on which many applications can be developed using a wide range of free tools.

FreeSWITCH was originally designed and implemented by Anthony Minessale II 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. We support various communication technologies such as Skype, SIP, H.323 and WebRTC making it easy to interface with other open source PBX systems such as sipXecs, Call Weaver, Bayonne, YATE or Asterisk.

FreeSWITCH can perform full video transcoding and MCU functionality using its conferencing module. 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 in mono or stereo 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. FreeSWITCH supports FAX, both over audio and T.38, and can gateway between the two. The developers are heavily involved in open source and have donated code and other resources to other telephony projects including openSER, sipXecs, The Asterisk Open Source PBX and Call Weaver. a Spec Sheet is available on our Confluence.

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FreeSWITCH 1.6.17 released!

Releases -

The FreeSWITCH 1.6.17 release is here!

This is just a routine maintenance release, but there are some great features in this release as well.

Release files are located here:

features New features and improvements that were added:

  • FS-10210 [mod_console] Added support for uuid config param and 'console uuid' api command to make reading logs from FreeSWITCH easier while running it in containers that read logs from stdout.
  • FS-10126 [core] General Video Improvements

build Improvements in build system, cross platform support, and packaging:

  • FS-10153 [build] Fixed mod_http_cache build on FreeBSD

bugs The following bugs were squashed:

  • FS-10220 [mod_conference] Fixed an issue with conference channel parameters not working
  • FS-10099 [mod_conference] Fixed a rare segfault on race condition when shutting down a conference
  • FS-10225 [mod_conference] Fixed an off-by-one error on the calculation of the correct layout to use when choosing a layout group while a file is playing.
  • FS-10059 [sofia-sip] Added a fix to correctly handle re-invites during t.38 call
  • FS-9765 [mod_sofia] Fixed to use switch_channel_var_true instead of switch_channel_get_variable to prevent allocating on every hold/unhold just to check if this is enabled
  • FS-10117 [mod_rayo] Fixed it to allow duplicate rayo signal-type configs for call progress detector to prevent segfaults
  • FS-10100 [mod_av] Fixed a crash on allocation error and other error cases when opening a file
  • FS-10209 [core][mod_av] Add auth_username and auth_password to media params for authentication when using streams
  • FS-10150 [core] Reduce writes to closed ssl sockets
  • FS-10195 [core]  Fixed intermittent segfaults 
  • FS-10222 [core] Added disable_audio_jb_during_passthru and disable_video_jb_during_passthru default behavior leaves the jitter buffer on for audio and video respectively during passthru situations unless the new variables are set to true.
  • FS-10233 [mod_local_stream] Fixed a segfault caused by trying to read a music file that is not open while playing a chime

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Development: 1.9.0

License: MPL 1.1

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