FreeSWITCH
FreeSWITCH is an open source telephony platform designed to
facilitate the creation of voice and chat driven products scaling from
a soft-phone up to a soft-switch. It can be used as a simple
switching engine, a media gateway or a media server to host IVR
applications using simple scripts or XML to control the
callflow.
The software supports various communication technologies such
as SIP, H.323, IAX2 and GoogleTalk making it easy to interface with
other open source PBX systems such as sipX,
OpenPBX, Bayonne, YATE or
Asterisk.
FreeSWITCH also support both wide and narrow band codecs
making it a solution to bridge legacy devices to the future.
The voice channels and the conference bridge module all can operate
at 8, 16 or 32 kilohertz and can bridge channels of different
rates.
It is capable of being embedded into other projects, as well
as being used as a stand-alone application.
Features include:
- Centralized User/Domain Directory (directory.xml)
- Nano Second CDR granularity
- Call recording (In Stereo caller/callee left/right)
- High Performance Multi-Threaded Core engine
- Configuration via CURL to your http server (xml_curl).
- XML Config files for easy parsing.
- Protocol Agnostic
- Configurable RFC2833 Payload type
- Inband DTMF generation and detection.
- Software based Conference (no hardware requirement)
- Wideband Conferencing
- Media / No Media modes
- Proper ENUM/ISN dialing built in
- Detailed CDR in XML
- Radius CDR
- Subscription server
- Shared Line Appearances
- Bridged Line Appearances
- Enterprise/Carrier grade Eventing Engine. (XML Events, Name
Value Events, Multicast Events)
- Loadable File formats and streaming
- Stream to Shoutcast
- Multi-lingual Speech Phrase Interface
- ASR/TTS support (native and via MRCP)
- Basic IP/PBX features
- Automated Attendant
- Custom Ring Back Tones
- XML RPC support
- Multiple format CDR's supported
- SQL Engine provides session persistence
- Thread Isolation
- Parallel Hunting
- Serial Hunting
- Support
- Paid support available
- Free support via IRC & e-mail
- Many supported codecs:
- CELT (32kHz ahd 48kHz)
- G.722.1 (wideband)
- G.722.1C (wideband 32kHz)
- G.722 (wideband)
- G.711
- G.726 (16k,24k,32k,48k) AAL2 and RFC3551
- G.723.1 (passthru)
- G.729 (passthru)
- AMR (passthru)
- iLBC
- speex (narrow and wideband)
- lpc10
- DVI4 (ADPCM) 8khz and 16khz
Return
to VoIP Home Page
Last Updated Tuesday, April 09 2013 @ 02:03 PM EDT |