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Possible Uses

  • Rating & Routing Server
  • Transcoding B2BUA
  • IVR & Announcement Server
  • Conference Server
  • Voicemail Server
  • SBC (Session Border Controller)
  • Basic Topology Hiding Session Border Controller
  • DAHDI, Khomp, PIKA, Rhino, Sangoma and Xorcom Hardware Support
  • Fax server
    • T.38 gateway, termination, and origination mode
    • T.30 to T.38 and T.38 to T.30 gateway
    • See also: mod_spandsp
  • And, of course, a PBX

Features

  • WebRTC support
  • 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 (mod_xml_curl).
  • XML Config files for easy parsing.
  • Protocol Agnostic
  • ZRTP support for transparent RTP based key exchange and encryption
  • Configurable RFC 2833 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 and play from Shoutcast and Icecast
  • Multi-lingual Speech Phrase Interface
  • ASR/TTS support (native and via MRCP)
  • Basic IP/PBX features
  • Automated Attendant
  • Custom Ring Back Tones (Early Media)
  • XML-RPC support
  • Multiple format CDRs supported
  • SQL Engine provides session persistence
  • Thread Isolation
  • Parallel Hunting
  • Serial Hunting
  • Mozilla Public License
  • Support
    • Paid support available
    • Free support via IRC & E-mail
  • Many supported codecs
    • CELT (32 kHz ahd 48 kHz)
    • G.722.1 (wideband)
    • G.722.1C (wideband 32 kHz)
    • G.722 (wideband)
    • G.711
    • G.726 (16k, 24k, 32k, 48k) AAL2 and RFC 3551
    • G.723.1 (passthrough)
    • G.729AB (Requires a license unless using passthrough)
    • AMR (passthrough)
    • iLBC
    • Speex (narrow and wideband) with RFC 5574 fmtp support
    • LPC-10
    • DVI4 (ADPCM) 8 kHz and 16 kHz
    • SILK
    • OPUS - RFC 6716
    • Video Codecs (passthrough):
      • Theora
      • H.261
      • H.263
      • H.264
      • MP4
    • See also: codecs
  • Live Migration of calls from one FreeSWITCH box to another. See Freeswitch_HA

Applications

  • Voicemail
    • Multitenancy - Enterprise/Carrier configuration
    • Time of Day Greetings
    • Urgent Message Tagging
    • E-mail Delivery
    • Playback and Rerecord messages before delivery.
    • Keys are templates so you can rearrange to fit your needs.
    • Callback support from inside voicemail.
    • Podcast of Voicemail (RSS)
    • Message Waiting Indicator (MWI)
  • Support for Queues (via mod_fifo or mod_callcenter)
  • Parking (via mod_fifo)
  • Conference
    • Software based Conferencing without any hardware requirements.
    • Wideband conferences.
    • Multiple on-demand or scheduled conferences with entry/exit announcements
    • Play files into the conference or a single member.
    • Relationships
    • TTS integration
    • Transfers
    • Outbound Calling
    • Configurable Key Lay
    • Volume, Gain and Energy level per call.
    • Bridge to Conference transition
    • Multi Party outbound dialing.
    • RFC 4579 SIP CC Conferencing for UAs
    • Automatic or on-demand recording
  • RSS Reader
  • Fax endpoint, gateway and passthrough mode.
    • T.30 (G.711) Audio Fax (via mod_spandsp) formerly known as mod_fax.
    • T.38 faxing (gateway, endpoint and passthrough)

Protocols

  • SIP with mod_sofia
    • UDP, TCP, SCTP and TLS transports for full SIP compliance.
    • SIP v.2.0 (RFC 3261)
    • IPv6 Support
    • SIP Session timers
    • RTP Timers
    • RFC 3263 (SRV and NAPTR)
    • RFC 3325
    • RFC 4694
    • SRTP via SDES (Works with Polycom, Snom, Linksys and Grandstream)
    • Blind SIP Registration
    • STUN Support
    • Jitter buffer
    • NAT Support
    • Distributed SIP registrations
    • Late Codec Negotiation
    • Multiple SIP registrations per user account.
    • Multitenancy - Multiple SIP UAs
    • SIP Reinvites.
    • Can act as an SBC (Session Border Controller)
    • Manage Presence
    • SIP/SIMPLE (can gateway to other chat protocols)
    • SIP Multicast Paging support for Linksys and Snom
    • Intercom/AutoAnswer support.
    • Call features like Call Hold (Re-INVITE), Blind Transfer (REFER), Call Forward (302), etc.

Languages

  • JavaScript (Using the Google V8 JavaScript engine.)
    • ODBC Support from inside your JavaScript
    • Extendable modules for JavaScript
    • Tone Generation
  • Ruby
  • Python
  • Perl
  • Lua

Cross Platform

  • Builds native on Windows in MSVC
  • Builds on Mac OS X, Linux, Solaris and *BSD.

Minimum/Recommended System Requirements

  • 32-bit OS (64-bit recommended)
  • 512MB RAM (1GB recommended)
  • 50MB of Disk Space

System requirements depend on your deployment needs. We recommend you plan for 50% duty cycle.

Performance

  • Tested under load for over 100 hours
  • 10,000,000+ calls
  • At rates exceeding 50 CPS

Performance will vary depending on application. You will need to test for your particular situation.

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