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.
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trixbox CEは、インストールが容易なAsterrisk PBXベースのVOIP電話システムです。trixboxは、自宅やオフィスで使用する目的で設計されています。trixbox CEにはCentOS Linux、Mysql、ビジネス品質の電話システムに必要な各種ツールが含まれています。
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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.
baresip is a bare-bones SIP user agent. It supports SIP, SDP, RTP/RTCP, and STUN/TURN/ICE, and IPv4 and IPv6, and is RFC-compliant and has portable C89 and C99 source code. A modular plugin architecture provides stdio, cons, and evdev user interfaces, celt, g711, g722, gsm, ilbc, l16, and speex audio codecs, alsa, coreaudio, gst, portaudio, oss, winwav, and mda audio drivers, speex_pp, speex_aec, speex_resamp, and sndfile audio filters, the avcodec video codec, avformat, quicktime, qtcapture, v4l, and v4l2 video sources, sdl, opengl, and x11 video display drivers, and srtp media encoding.
VoiceOne is a Linux distribution that gives you the ability to install a PBX platform with an easy to use Web-based GUI. It also provides a framework for building a communication server adding various plugins. Its main features are Asterisk 1.8 with realtime configuration with MySQL, a Ubuntu 10.04 base, and support for both hard disk and Compact Flash card installation.
libre is a generic library for real-time communications with asynchronous I/O support. It is written in portable POSIX source code that conforms to the ANSI C89 and ISO C99 standards. It is robust and fast, with a low memory footprint. It also features RFC compliance and support for IPv4 and IPv6. Protocol implementations include SIP, SDP, RTP/RTCP, BFCP, DNS, STUN/TURN/ICE, HTTP, and WebSockets.
Asterisk speech recognition is an AGI script that makes use of the Google voice recognition engine in order to render speech to text and return it back to the dialplan as an asterisk channel variable.
Real time communication software built to provide face-to-face advantages to remote gamers.
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EMIPLIB is a library to facilitate the development of programs that need to stream several kinds of media over IP. The library consists of several kinds of components that can be linked together in various ways, thereby providing a flexible framework. It also provides some ready-to-use classes for the transmission of audio and video over IP. Streams originating from the same participant can be synchronized.
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SEMS is a media and application server for SIP based VoIP services. It shows good performance doing basic services like announcements and conference for combination with external application servers. Thanks to its easy-to-use and flexible application development framework and back-to-back user agent support, application logic and media serving can be combined in the same process. Basic applications like announcement, pre-call announcement, RBT, conference, voicemail, mailbox, and lots of example applications are available. Scripting can be done in Python and a simple state machine description language. Support All commonly used free codecs (including g711, gsm, iLBC, speex, adpcm, and l16) are supported. Other features include wideband, ZRTP encryption, a SIP registrar client, an XMLRPC server/client, and a DIAMETER client.
Flite For Asterisk provides the "Flite" dialplan application, which allows you to use the Flite TTS Engine with Asterisk. This module invokes the Flite TTS engine locally, and uses it to render text to speech. It supports voice selection and 8kHz or 16kHz sample rates to provide the best possible sound quality along with the use of wideband codecs. It works with Asterisk 1.6.x, 1.8, and 10.
Text translation for Asterisk using MS Translator uses the Microsoft Translator API to translate text strings or detect their language and return them as Asterisk channel variables.
Scopserv Communicator is an audio/video Internet phone and instant messenger that supports some of the most popular VoIP and instant messaging protocols such as SIP, Jabber (XMPP), AIM/ICQ, MSN, Yahoo! Messenger, and a whole lot of other useful features. ScopServ Communicator is based on the SIP Communicator softphone.
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restund is a modular STUN/TURN server that is designed around the principle of a lightweight core and server modules that extend its functionality. Both UDP and TCP are supported, along with IPv6 and IPv4. Supported modules include STUN, TURN, MySQL database, syslog, and status monitoring.