Internet Phone Software

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Internet Phone Software

  • Monitor your Cisco ASA like an expert Monitor your Cisco ASA like an expert Icon
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  • Elastix Icon

    Elastix

    Unified Communications Server

    Elastix is a software-based PBX powered by 3CX and based on Debian. An open-standards solution, Elastix is an easy to install and manage UC system compatible with popular IP phones, gateways and SIP trunks. Elastix is complete with unified communications features such as integrated WebRTC video conferencing, chat, presence and softphones and smartphone clients for Windows, Mac, iOS and Android.

  • trixbox

    trixbox CE is an easy to install, VOIP phone system based on the Asterisk PBX. trixbox is designed for home or office use. trixbox CE includes CentOS linux, mysql, and all the tools needed to run a business quality phone system. (formerly asterisk@home)

  • Mumble Icon

    Mumble

    Low-latency, high quality voice chat for gamers

    Mumble is an open source, low-latency, high quality voice chat software primarily intended for use while gaming. It includes game linking, so voice from other players comes from the direction of their characters, and has echo cancellation so the sound from your loudspeakers won't be audible to other players.

  • sipp

    Sipp is a performance testing tool for the SIP protocol. Its main features are basic SIPStone scenarios, TCP/UDP transport, customizable (XML-based) scenarios, dynamic adjustment of call-rate and a comprehensive set of real-time statistics.

  • libsrtp

    The libSRTP project has been moved to https://github.com/cisco/libsrtp

  • Identify and respond to threats before damage is done Identify and respond to threats before damage is done Icon
    Identify and respond to threats before damage is done Icon

    A SIEM that makes it easy to use logs for security, compliance, and troubleshooting.

    SolarWinds® Log & Event Manager (LEM) delivers comprehensive Security Information and Event Management (SIEM) capabilities in a highly affordable, easy-to-use, and quick-to-deploy virtual appliance. LEM provides the visibility, security, and control that IT professionals need to alert on and respond to security threats, and assist in meeting the regulatory compliance guidelines of HIPAA, PCI DSS, SOX, and more. Try it FREE for 30 days!
  • STUN Client and Server

    This project implements a simple STUN server and client on Windows, Linux, and Solaris. The STUN protocol (Simple Traversal of UDP through NATs) is described in the IETF RFC 3489, available at http://www.ietf.org/rfc/rfc3489.txt

  • Disc-OS

    Vem ai o Disc-OS 3.0 Aguardem.. Modificações: Asterisk 14 ou Asterisk 15 Beta-1 Nova Interface UI Ubuntu 17.04 (Zesty Zapus) PHP 7.1 Disc-OS é uma distribuição de um PABX IP baseado em software livre. Desenvolvido para o mercado brasileiro com interfaces em português, de fácil instalação e configuração, contendo Linux customizado, software Asterisk 1.4 e configurador Disc.

  • jphonelite Icon

    jphonelite

    Java VoIP Softphone (SIP)

    jphonelite is a Java SIP VoIP SoftPhone for Desktops (Windows, Linux, Mac) and Android. Features 6 lines with transfer, hold, conference (up to all 6 lines), g711 u/a, g722, g729a, and video (video support in Linux or Windows only and includes H263/H264/VP8). Applet includes full JavaScript support. STUN/TURN/ICE supported. Encrypt media with SRTP. DTLS Key Exchange. Video conferencing with jPBXLite/0.12+ Please ignore the malware warnings in downloads. It's a false positive.

  • Skype Audio Player

    Skype Audio Player allows playing audio files (mp3, wav, wma) during Skype calls to both parties. It is a simple tool without luxurious functions, useful for those who study foreign languages using Skype and need to play sample dialogs or audio tests.

  • Java SIP softphone Icon

    Java SIP softphone

    Peers is a very simple softphone. It's a SIP User-Agent, written in java, it works on windows, linux and mac. It can be used with SIP servers like opensips or asterisk IPBX. It supports G711 codec (PCMU and PCMA) and telephone-events (DTMF).

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  • OpalVOIP

    Open Phone Abstraction Library (OPAL) is a C++ multi-platform, multi-protocol library for Fax, Video & Voice over IP and other networks. Also included is the Portable Tool Library (PTLib) which is a C++ multi-platform abstraction library and collection o

  • Activa for Asterisk

    Activa brings the Asterisk IP PBX to the call center. Built on top of Asterisk, Activa components enable successful call center implementations adding value in areas such as computer telephony, screenpop&click2dial, agent control, automatic dialing...

  • Chan-SCCP channel driver for Asterisk Icon

    Chan-SCCP channel driver for Asterisk

    Replacement for the SCCP channel driver in Asterisk. Extended features include Shared Lines, Presence / BLF, customizable Feature Buttons, and Custom Device State. Visit our discussion mailing list for help and join us as a developer if you like.

  • Kiax - an IAX Softphone

    Kiax is a softphone (soft phone, VoIP client) with a simple and comfortable user interface for making VoIP calls to Asterisk PBX. It depends on the iaxclient library to use Asterisk's IAX2 protocol for easy call configuration and audio setup.

  • OSDial Icon

    OSDial

    OpenSource Call Center and Dialer System

    An Open Source (predictive) Dialer. OSDial is a full featured GPL dialer which does Manual, Power and Predictive dialing out of the box. Its easily installed through one of our .iso images. Install manual available at http://store.osdial.com/ Full details available at web site (http://osdial.com).

  • OpenH323

    This project has been superseded by OpalVoip (https://sourceforge.net/projects/opalvoip/) and H323Plus (https://sourceforge.net/projects/h323plus/) The OpenH323 project provides full featured, interoperable, Open Source implementation of the ITU H.323 teleconferencing protocol that can be used by personal developers and commercial users without charge.

  • Sofia-SIP

    Sofia-SIP - a RFC3261 compliant SIP User-Agent library.

    Downloads: 117 This Week Last Update: See Project
  • siptapi

    A TAPI driver for SIP. SIPTAPI gives you a click2dial feature with any TAPI enabled application (e.g. MS Outlook) and SIP PBX/proxy. This is the "free" version of SIPTAPI. There is also an enhanced commercial version available at www.ipcom.at.

  • XMeeting - Video Conferencing for OS X

    This project develops video conferencing and VoIP applications for Mac OS X. XMeeting is based on the work of the OpenH323 project which provides the libraries for H.323 and SIP support.

  • wxCommunicator

    wxCommunicator is a cross platform SIP softphone written in C++ utilizing customized sipXtapi user agent library and wxWidgets 2.8.9 GUI library. For a list of supported features see http://wxcommunicator.sourceforge.net/features.html .

  • UCT IMS Client

    Open Source 3GPP IMS Client

  • VMukti - IP Communications Suite

    VMukti open source is leading Asterisk/ Yate enabled p2p Video IP Communications Suite for Web / PSTN. These serverless broadband ready platform enable OS community to save 90% on capital & operating costs over proprietary software for conferencing &

  • IPP Codecs

    Intel Integrated Performace Primitives audio/video codecs plug-in for the OPAL/OpenH323 library including G.728, G.729, G.723.1, G.722.2 GSM-FR, GSM-AMR, H.261, H.263, H.264 and MPEG4 part 2.

  • GNU Gatekeeper (GnuGk) Icon

    GNU Gatekeeper (GnuGk)

    H.323 Gatekeeper for VoIP and videconferencing

    The project has moved! Please find current versions at https://www.gnugk.org/ The GNU Gatekeeper (GnuGk) is a full featured H.323 gatekeeper under GPL license. It supports VoIP and videoconferencing and includes Radius and database support and many call routing functions. The project has moved! Please find current versions at https://www.gnugk.org/

  • asterCRM

    asterCRM is a call center software for asterisk based VoIP system, also it has some CRM functions. It provide useful features such as pop-up, predictive dialer, click to call, extension status .... astercrm could work with all asterisk based system.

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