Search Results for "sofia-sip"
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Showing 36 open source projects for "sofia-sip"
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jfPhone
VoIP/SIP SoftPhone (formerly known as jphonelite)
jfPhone is a VoIP/SIP SoftPhone for Desktops. -
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A C++ framework utilizing Design Patterns for creating Linux and Windows communications applications that contain Dialogic® products. Includes media and network classes (analog, digital, SIP, H323), multithreaded event handling, distributed app support.
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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. -
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jphonelite
Java VoIP Softphone (SIP) (replaced by jfPhone)
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. -
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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 .
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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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PartiSIPation is a sip user agent which has an easy replaceable gui. So it can be used as a softphone as well as it can be integrated in any kind of application, e.g. online games.
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dtmfbox is a small softswitch application (SIP/CAPI), that can be used to control different tasks over telephone keyboard via DTMF. Mostly, it was made to run on the AVM FRITZ!Box 7170 (mipsel) but works under Unix/Linux and Windows, too.
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A Sip live audio feeding agent. The agent captures audio from sound card and sends live audio stream(uLaw) to caller(sip phone) using RTP. It is based on Peers 0.3(http://peers.sourceforge.net/). Can be used in IP telephony to broadcast live audio.
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Prototype testbed implementation of the IETF Media Server Control (MEDIACTRL) SIP Control Framework, comprehensive of both control and processing functionality (as in IMS MRF, Media Resource Function).
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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.
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VoIPER is a VoIP security testing toolkit incorporating several VoIP fuzzers and auxilliary tools to assist the auditor. It can currently generate over 200,000 SIP tests and H.323/IAX modules are in development. It's also a damn cool project name ;)
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FreeRADIUS module providing connectivity to CDRTool prepaid engine. Provides prepaid authentication for calls proxied by OpenSER and returns to OpenSER MaxCallDuration and user Credit. Can be easily extended to support other SIP or H323 devices.
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VoIP Toolkit / Call Control with Integrated Media. High-level Java API for creating SIP enabled VoIP applications. Suitable for either the desktop (softphone, phone applet, incoming call gatekeeper) or server-side (auto attendant, ACD, voicemail).
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This project contains a SIP stack and server applications built on top of the stack, examples of which are: Stateless Proxy, Registrar,NAT Keepalives,MWI Notifier server. The server applications have been designed to work in association with Asterisk.
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With SIP Proxy you will have the opportunity to eavesdrop and manipulate SIP traffic. Furthermore, predefined security test cases can be executed to find weak spots in VoIP devices. Security analysts can add and execute custom test cases.
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HATEFNA هاتفنا .....http://www.hatefna.org ..... (PC&Mobile)IP SoftPhone over SIP and ASTERISK using JAVA for PC and Symbian C++ for Symbian Mobiles Phones ..... برمجيات للمهاتفة ببروتوكول الإنترنت
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SIPStir is a bulk call generator for SIP devices/servers. The server piece is capable of generating an infinite variety of call scenarios based on a simple scripting language. A wxPython client can connect to the server to monitor and modify progress.
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Command-line perl script to view PortaSIP SIP server log files in convenient form. It also may be used for other SIP log formats after small changes.
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nSIP is an RFC 3261 compliant implementation of SIP (Session Initiation Protocol) for .NET, written in C#. nSDP is an RFC 4566 compliant SDP parser also in C#, and can be used to make payloads to nSIP messages or to parse any SDP message.
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'The Genie' is a Java based SIP User Agent that can talk to any other SIP User Agent via SIP Proxy or Asterisk. Also Planned: 1-click sign-in to Yahoo or any other site by storing hashes of user/passwd pairs in a 'Wallet' and they can be accessed onl