Search Results for "linux gpib support"
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Showing 24 open source projects for "linux gpib support"
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TelcoChisel OS
Advanced Telecom Security Live Linux OS for 5G/4G Cellular Auditing, S
TelcoChisel is an advanced, bootable live Linux operating system purpose-built for cellular security researchers, Software Defined Radio (SDR) engineers, and telecommunications penetration testers. Eliminating dependency conflicts and driver compilation issues, TelcoChisel delivers an out-of-the-box, carrier-grade laboratory equipped with 88 pre-configured telecommunications security tools across 10 specialized domains. Built on Ubuntu 24.04 LTS (Noble Numbat) with the Linux... -
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CSTA Inside
Open source CSTA library
Aim of this project is to provide a library for C/C++ developers to use ECMA-269/ECMA-323 CSTAIII XML protocol. It will also include: CSTA interface for FreeSWITCH and example client application. (extending the PBX and protocol support is in progress). Check wiki for details. -
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BlackBelt Privacy- Tor i2p WASTE VidVoIP
Browse on Tor / i2p - Anon p2p Chat / FileTx, Conf / Video VoIP
...An installer, for : Vista 32/64, Win7 32/64, Win8 32/64, Win10 32/64, Win11, Linux (Wine) -
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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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Stuntman - STUN server and client
High performance, production quality STUN server and client library
New version 1.2. This is the code to STUNTMAN - an open source STUN server and client code by john selbie. Compliant with the latest RFCs including 5389, 5769, and 5780. Also includes backwards compatibility for RFC 3489. ICE and WebRTC ready. Version 1.2 compiles on Linux, MacOS, BSD, and Solaris. Supports the STUN protocol on both UDP and TCP for both IPv4 and IPv6. Windows binaries are also provided. Additional features are in development. This is a mirror of the code on... -
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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/ -
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MCU Media Server
SIP Video Multiconference Media Server with WebRTC support.
REPOSITORY MOVED TO GITHUB!! https://github.com/medooze/media-server Video Multiconference Media Server with WebRTC support. Provide Multiconference and video broadcasting services to any SIP service. Supports VP8, H264, MP4V-ES, H263 and H263P, continuous presence, RTMP flash broadcasting, adhoc conferences, load balancing and administrative WEB interface. JSR309 driver implementation under development. . -
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mod_MumbleLink
Positional Audio Communication for Minecraft with Mumble
A Mod so that Minecraft now natively supports Mumble's positional audio feature. This means: Directional and positionally attenuated VOIP in relation to the game world. Please visit the Forum for information about the newest Version! Main Forum-Thread: http://www.minecraftforum.net/forums/mapping-and-modding/minecraft-mods/1272675 TheSkorm's Fork on GitHub: https://github.com/TheSkorm/mod_mumblelink Mumble: http://mumble.sourceforge.net Donations are greatly appreciated:... -
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sigtran-smsc
PoC: Short Message Service Center (SMSC) over SIGTRAN
...As being developed in very limited schedule (bachelor thesis work) it was limited to have very basic functionality: receiving message and deliver it immediately as-is to the receiver. There is no message caching nor different message charset support. It uses following protocol stack: MAP (moForwardSM and mtForwardSM) TCAP (Begin with Invoke + End with ReturnResultLast) SUA (SCCP User Adaptation): ASP UP/ASP ACTIVE + ACK's and CLDT messages only SCTP (Linux Kernel - SCTP implementation) IP Current implementation is tested in SAMK NGN [http://www.samk.fi/ngn] laboratory's 3G mobile network (real Nokia MSC/VLR network hardware with UTRAN RLC and node-B). ... -
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A Easy 3G dialer. A dial system with support for most current GPRS and 3G phones. Simple, the idea is to detect, connect, use and disconnect . Languages: pyQT4, Shell Script.
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Video Applications for Asterisk (libh43m, app_h324m, app_mp4, app_rtsp, app_transcode, app_swf) H324m stack, mp4 record and playblack, rtsp support, video transcoding and swf file playback.
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What is t38modem? From your application view point it's a fax/voice modem pool. From IP network view point it's a H.323/SIP endpoint with T.38 fax support. From your view point it's a gateway between an application and IP network. Works with HylaFAX.
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OpenAL mixer aims to provide a multiplatform API built on top of OpenAL to get/set volume values (including support for capture devices) and list, select and mute input and output lines.
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BladeWareVXML is a portable VoiceXML 2.1 interpreter that is an enhanced version (performance, usability and integration) of OpenVXI. A commercial version, with documentation, sample code, and support options, is available from the Commetrex Website.
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IM,WebIM and Sip voip tool. Client is Win32, Server is CentOS 5. support video,audio, file trans, sip call.
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Anca is a H.323 client for Qt 3.x. It means that it allows communicating (voice/video) with other similar clients over internet, eg. Gnomemeeting. Its design is minimalist and clean; has a support for embedded devices. It is very modular.
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PublicVoiceXML is a VoiceXML 2.0 browser, designed to work also on low cost telephony hardware using DTMF navigation with hooks to 3rd party TTS Modules. VoIP support is under development. Tech support and sample applications available.
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Isdnserver is intended to be an ISDN-server-suite with built-in decoding, tracing, tracking, voice box, and telephony support. It works with the isdn4linux drivers. Capi2.0 support will be added in the future.
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java telephony API, Dialogic's hardware support, telephony application IDE, mutlichannel telephony applications Server
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A wxWindows based IAX client utilizing the phonecore module with support for Digium's S100U/P and Quicknet PhoneJack, along with just a plain ole regular sound card.
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Siphon will ultimately be a Software Voice-Over-IP phone using the SIP protocol. It will support quicknet cards as well as traditionnal soundcards. It is developped for Linux (gtk) and the win32 platform.
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Open Telephony Server or OtServer for short it's a set of tools for developing telephony applications. A CallFlow Graphical Editor. Dialogic support.
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sippet
C++ SIP stack based on Chrome source code
This C++ library has been designed as a Chrome SIP stack. Sippet is an open-source SIP User-Agent library, compliant with the IETF RFC 3261 specification. It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and P2P communication services. The main target was to enable Javascript applications to use UDP, TCP and TLS transports along WebSocket. Existing SIP solutions for the browser are forced to use the WebSockets API to...
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