46 projects tagged "voip"
Updated 21 Feb 2013
Ekiga
| Pop | 695.54 |
|---|---|
| Vit | 45.30 |
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.
Updated 07 Jan 2013
Yet Another Telephony Engine
| Pop | 423.93 |
|---|---|
| Vit | 27.08 |
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.
Updated 10 Sep 2011
VoiceOne
| Pop | 243.61 |
|---|---|
| Vit | 10.01 |
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.
Updated 18 Jul 2012
EMIPLIB
| Pop | 95.55 |
|---|---|
| Vit | 6.64 |
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.
Updated 22 Apr 2009
sware
| Pop | 25.77 |
|---|---|
| Vit | 2.07 |
sware is a Perl script to connect a SNOM 3x0 telephone to the databases of eGroupWare, Open-Xchange, and vtiger to display addressbook and calender entries on your SNOM telephone. It uses LDAP or MySQL authentication. Incoming calls are shown as a lastname or company name in the display if the result is available in the database.
Updated 08 Jan 2012
SIPFwd
| Pop | 62.53 |
|---|---|
| Vit | 4.75 |
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.
Updated 10 Apr 2009
LOUD_LISTENER
| Pop | 35.16 |
|---|---|
| Vit | 38.82 |
LOUD_LISTENER is a light-weight, high-speed voice packet recorder that can target a particular call based on a SIP URL or TG, designed to satisfy regulatory lawful Intercept requirements.
Updated 06 Apr 2011
Voip Development Kit (VDK)
| Pop | 62.10 |
|---|---|
| Vit | 3.40 |
VoIP Development Kit (VDK) is a cross-platform framework that allows creation of VoIP applications in a very rapid and easy way using VirtualDev technology. Very little knowledge (if any) of VoIP protocols is needed to create your first application.
Updated 16 Oct 2009
Imptalk
| Pop | 57.84 |
|---|---|
| Vit | 1.51 |
Real time communication software built to provide face-to-face advantages to remote gamers.
Updated 11 Nov 2011
Asterisk Flite
| Pop | 77.59 |
|---|---|
| Vit | 3.12 |
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.
