FreeSwitch is a fully combined package to build everything from softphones to full Class 5 switches. You can embed FreeSWITCH into a soft- (or hard-) phone or OpenWRT router, hook it to PRI circuits through FreeTDM and hardware interfaces such as the Sangoma A100 series, or use it to build an office PBX phone system — from scratch, or using packaged add-ons like FusionPBX.

• A series of resources leveraged by the Apache webserver.
• A lightweight implementation of the SQL database engine.
• A multi-tenant platform where each tenant (customer) is segregated.
• Increased capabilities for maximizing concurrent calls, given the same underlying hardware.

• 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)
• Support WebRTC
• Text-to-speech and Automatic Speech Recognition support

• Multi-level encryption support such as WSS, HTTPS, SRTP, SIPS, SSL, and more
• mod_skinny – Skinny Call Control Protocol (SCCP)
• mod_verto – VERTO WebRTC Signaling Protocol
• SIP with mod_sofia
  • UDP, TCP, SCTP and TLS transports for full SIP compliance.
  • xiWS and WSS transport for full WebRTC compliance.
  • SIP v.2.0 (RFC 3261)
  • IPv6 Support
  • SIP Session timers and RTP Timers
  • RFC 3263 (SRV and NAPTR), RFC 3325, and RFC 4694
  • SRTP via SDES (Works with Polycom, Snom, Linksys, and Grandstream)
  • Blind SIP Registration and SIP Reinvites.
  • STUN Support, Jitter buffer, and NAT Support
  • Distributed SIP registrations and Late Codec Negotiation
  • Multiple SIP registrations per user account.
  • Multitenancy – Multiple SIP UAs

• Software-based wideband conferencing without any hardware requirements.
• Multiple on-demand or scheduled conferences with entry/exit announcements.
• Play files into the conference or a single member.
• Relationships, TTS integration, Transfers, and Outbound Calling.
• Configurable Key Lay.
• Volume, Gain, and Energy level per call.
• Bridge to Conference transition
• MultiParty outbound dialing.
• RFC 4579 SIP CC Conferencing for UAs.
• Automatic or on-demand recording.

• Support for an unlimited number of customer accounts.
• Support a mix of Post-paid and Prepaid accounts.
• Both real-time billing (RADIUS) and offline billing (CDR Files).
• Calls can be posted to 3rd party applications such as OSCommerce, Optigold, or AgileBill.
• Least Cost & Failover Routing.
• Dynamically map incoming DIDs to the correct user via the web interface
• Impose credit controls on incoming DIDs.