51 projects tagged "voip"

Download Website Updated 17 Mar 2014 Yet Another Telephony Engine

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Pop 681.21
Vit 41.37

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.

Download Website Updated 21 Feb 2013 Ekiga

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Pop 420.47
Vit 21.05

Ekiga (formely known as GnomeMeeting) is a soft phone, video conferencing, and instant messenger application for use over the Internet. It supports HD sound quality and video up to DVD size and quality. It is interoperable with many other standards compliant software, hardware, and service providers as it uses both of the major telephony standards, SIP and H.323.

Download No website Updated 19 Mar 2014 Zentyal

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Pop 353.41
Vit 14.47

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.

Download Website Updated 23 Jan 2014 baresip

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Pop 262.27
Vit 11.39

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.

Download Website Updated 12 Apr 2014 libre

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Pop 222.49
Vit 17.06

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.

Download Website Updated 10 Sep 2011 VoiceOne

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Pop 221.47
Vit 8.45

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.

Download Website Updated 02 Aug 2013 GROUP-E

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Pop 170.34
Vit 13.37

GROUP-E is collaboration software which integrates groupware, project management, and business server on one platform. The solution is based on a LAMP architecture (Linux, Apache, MySQL, PHP). GROUP-E offers project management, transparent Samba (file server) integration, integration of Cyrus IMAP server with administration and personal SIEVE filters, support for SyncML 1.0, LDAP-based user management with single sign-on authentication, and LDAP contact databases. GROUP-E also provides ActiveSync for synchronization.

No download Website Updated 18 Jul 2012 EMIPLIB

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Pop 89.48
Vit 5.68

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.

Download Website Updated 11 Nov 2011 Asterisk Flite

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Pop 72.97
Vit 2.73

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.

Download Website Updated 08 Jan 2012 SIPFwd

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Pop 72.39
Vit 3.94

The SIP forwarding daemon (implemented as a stateless SIP proxy) allows you to seamlessly forward SIP requests to other SIP servers. Its main purpose is to enable users to use their own domain name in SIP URIs without the hassle of having to run a full-blown SIP server (by forwarding SIP requests to third-party SIP servers). Configuration information is stored in an SQLite database, and low resource consumption is a main priority for the project.

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