- Home
- About Us
- Services
- Products
- Technology
- Information
- Online Shop
- Contact Us
goava VoIPLow cost Local and International Calls
 |
|
| Original author(s) | Anthony Minessale |
|---|---|
| Developer(s) | Many contributors |
| Stable release | 1.6.19 (July 13, 2017) [±] |
| Development status | Active |
| Written in | C |
| Operating system | Unix-like, Windows, Solaris, OS X |
| Platform | Cross-platform |
| Available in | multi-lingual |
| Type | VoIP software, Softswitch |
| License | Mozilla Public License (MPL) |
| Website | freeswitch |
FreeSWITCH is a free and open source application server for real-time communication, WebRTC, telecommunications, video and Voice over Internet Protocol (VoIP). Multiplatform, it runs on Linux, Windows, MacOS and FreeBSD. Is used to build PBX systems, IVR services, videoconferencing with chat and screen sharing, wholesale least cost routing, Session Border Controller (SBC) and embedded communication appliances. Has full support for encryption, ZRTP, DTLS, SIPS. Can act as a gateway between PSTN, SIP, WebRTC, and many other communication protocols. Its core library, libfreeswitch, can be embedded into other projects. It is licensed under the Mozilla Public License (MPL), a free software license.
The FreeSWITCH project was first announced in January 2006 at O'Reilly Media's ETEL Conference.[1] In June 2007, FreeSWITCH was selected by Truphone for use,[2] and in August 2007, Gaboogie announced that it selected FreeSWITCH as its conferencing platform.[3]
FreeSWITCH's first official 1.0.0 release (Phoenix) was on May 26, 2008.[4] A minor 1.0.1 patch release came out on July 24, 2008.[5] At ClueCon 2012 Anthony Minessale announced[6] the release of FreeSWITCH version 1.2.0[7] and that the FreeSWITCH development team had adopted separate stable (version 1.2) and development (version 1.3) branches.
FreeSWITCH 1.4, released at early 2014, is the first version support SIP over Websocket and WebRTC.
FreeSWITCH 1.6 added support for video transcoding and video conferencing, Verto protocol for WebRTC, and all WebRTC codecs and standards.
According to the lead designer, Anthony Minessale,[8] FreeSWITCH is intended to be a softswitch that is built on top of a solid core, driven by a state machine.[9] The stated goals of the project include stability, scalability, and abstraction.
To reduce complexity, FreeSWITCH uses freely available software libraries that perform needed functions. Some dependencies are:[10]
Not all of these software dependencies are required to build the core freeswitch application, but are dependencies of various external modules, such as codecs. FreeSWITCH is a modular application, in which modules can extend the functionality but the abstraction layer prevents inter-module dependency. The goal is to ensure that one module is not required to load another.[11]
FreeSWITCH includes many modules that provide many telephony applications, such as conferencing, XML-RPC control of live calls, interactive voice response (IVR), speech synthesis and speech recognition, public switched telephone network (PSTN) interfaces for analogue and digital circuits, voice over IP protocols including Session Initiation Protocol (SIP), Verto, Skinny Client Control Protocol (SCCP), H.323, Extensible Messaging and Presence Protocol (XMPP), GoogleTalk, T.38 and others.[12]
FreeSWITCH provides an application programming interface that exposes primitives for call control and IVR functionality. Applications may be written in the C language, C++, Python, Perl, Lua, JavaScript, Java and Microsoft .NET via Microsoft's CLR or via Mono.[13]
Call control applications can use the Event Socket, which is an Internet socket-based communications facility within FreeSWITCH providing a language independent interface. The Event Socket Library (ESL) and "ESL-wrappers" are available for Erlang, JavaScript, Lua, Perl, PHP, Python, and Ruby.
As of FreeSWITCH version 1.4, support exists for WebRTC.[14]
In FreeSWITCH 1.6 support was added for Video muxing and complete WebRTC, wss, dtls, SIP.js, Verto.js, Opus 48khz to 8khz, resilient up to 40% packet loss.
Main FreeSWITCH 1.6 features:
[16] FreeSWITCH has always been a powerful platform for conferencing, starting many years ago as a hugely scalable audio conferencing bridge. In a breakthrough at ClueCon 2015 in Chicago Illinois, FreeSWITCH's creator Anthony Minessale II announced support for video transcoding, mixing, manipulation, and Multipoint Control Unit (MCU) functionality. FreeSWITCH now has the most advanced and mature video conferencing features:
FreeSWITCH supports a variety of audio and video codecs:[17]
G.723.1, H263 and H264 are supported in pass-through mode. Since the raw compressed data is passed through between callers without any processing, this allows support for some codecs that cannot be provided free of charge due to patent or other licensing issues.
The software supports hardware transcoding cards, such as produced by Sangoma.[19] These implement codecs in hardware, reducing the CPU usage of the server. Some of these codecs are fully licensed, providing an alternative to the pass-through options above.
FreeSWITCH occupies a space between pure switches that simply route calls, such as Kamailio and OpenSIPS, and those that provide primarily PBX or IVR functionality, such as Asterisk and its derivatives. FreeSWITCH provides building blocks from which applications – such as a PBX, a voicemail system, a conferencing system or a calling card – can be built using any of the supported languages.[22]
FreeSWITCH is a core component in many PBX in a box commercial products and open-source projects. Some of the commercial products are hardware and software bundles, for which the manufacturer supports and releases the software as open source.
