Software Map

OTRS is a platform independent Web-based help desk system that supports service organization of any kind (e.g. IT service, customer and technical product service, complaint management, public services, etc.) to increase their efficiency. It increases transparency as well as service quality and lowers your total cost of ownership. It has been certified ITIL V3 compatible by PinkVERIFY for incident, problem, change, service asset and configuration, request fulfillment, and knowledge management. Other ITIL processes like service catalog and service level management are supported as well.

The GNU Gatekeeper is a free H.323 gatekeeper based on the OpenH323 project. You can use it to manage a Voice-over-IP network and let endpoints (e.g., Netmeeting) communicate through symbolic names. It also has an external interface for billing and other applications. It runs on a number of Unix versions (including Linux and Solaris) and Windows.

Yate is a next-generation telephony engine. While currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended. It supports SIP, H.323, IAX, MGCP, Jingle, Jabber, E1, T1, analogic, robbed bit, ISDN PRI, BRI, and SS7. YateClient is an Instant Messenger and Voice application for Jabber/Jingle IM, SIP, H.323, and IAX VoIP protocols.

Asterisk is a hybrid TDM and packet voice PBX (Private Branch eXchange) and IVR platform with ACD functionality. It acts as middleware between the Internet (IAX, SIP, MGCP, Skinny, H.323), telephony channels (like Zaptel, T1, PRI, E1, FXO, FXS, VoIP, VoFR, ISDN, modems, Internet Phone Jack, etc.), and applications (like voice-mail, conferencing, directories, MP3 players, intercoms, etc.). It has many advanced features such as a codec translation API. The base distribution includes several channel backends, as well as applications. However, the beauty of Asterisk is its ability to be extended using its APIs, dynamic module loader, and AGI scripting interface. End users can even write their own applications that run on the system in C or any scripting language of their choice.

Linphone is an audio and video Internet phone with GTK+ and console interfaces. It uses the SIP protocol, and is compatible with most SIP clients and gateways. It can use various audio and video codecs such as Speex, GSM, G711, G722, ilbc, amr, Theora, H263-1998, MPEG4, H264, VP8, and snow.

SFLphone is an SIP/IAX2 compatible softphone. The goal is to create a robust enterprise-class desktop phone. While it can serve home users very well, it is designed for intensive corporate use.

Gammu (formerly known as MyGnokii2) is a cellular manager for various mobile phones/modems. It supports a wide variety of Nokia, Symbian, and AT devices (Siemens, Alcatel, Falcom, WaveCom, IPAQ, Samsung, SE, and others) over cables, infrared, or BlueTooth. It contains libraries with functions for ringtones, phonebook, SMS, logos, WAP, date/time, alarm, calls, and more (used by external applications like Wammu). It also includes a command line utility that can make many things (including backups) and an SMS gateway with full MySQL and PostgreSQL support from the PHP interface.

gnokii is a multisystem tool suite for mobile phones. It provides a library to communicate with a phone hiding the communication protocol. The library handles SMS, phonebook, calendar, phone calls, and other mobile phone capabilities. It supports Nokia-FBUS mobiles, AT-capable phones (most of the mobiles), as well as Symbian-based phones.

Kamailio (formerly OpenSER) is a high-performance SIP (RFC3261) server with a flexible architecture and many extensions. The server implements proxy, registrar, redirect, and location SIP/VoIP services. It has support for UDP, TCP, TLS, and SCTP transport layers, DNSsec, ENUM, AAA via database, RADIUS, DIAMETER, gateways to SMS and XMPP, least cost routing, load balancing, NAT traversal, and call processing language. Kamailio implements SIMPLE presence and instant messaging extensions, and includes an embedded XCAP server and MSRP relay. It can be also used as a routing SIP sever for WebRTC via WebSocket.

Zentyal Server aims at offering small and medium businesses (SMBs) a native drop-in replacement for Windows Small Business Server and Microsoft Exchange Server which can be set up in less than 30 minutes and is both easy to use and affordable.

Bayonne is the telephony server of the GNU project. It offers a script-driven threaded multi-line state event telephony service on GNU/Linux, xBSD, and Microsoft Windows for building voice response systems, and uses telephony plugins for runtime driver configuration. It also features "TGI" for making Perl applications "telephony aware". It may be used to build telephony-based system administration, home automation, automated attendant, v-commerce, and voice messaging systems.

Speex is a patent-free compression format designed especially for speech. It is specialized for voice communications at low bit-rates in the 2-45 kbps range. Possible applications include Voice over IP (VoIP), Internet audio streaming, audio books, and archiving of speech data (e.g. voice mail).

Sendpage sends alphanumeric messages to pagers or SMS phones. The server dials paging centrals or SMS centers, and communicates via the TAP or UCP protocol. Sendpage runs as an SNPP daemon with queue-management, multiple modems, and email notification.

OpenSIPS is a mature implementation of a SIP server/proxy. It is more than a SIP proxy/router, as it includes application-level functionalities. OpenSIPS, as a SIP server, can server as the core component of any SIP-based VoIP solution.