Search Results for "linux operating system"
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Showing 108 open source projects for "linux operating system"
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VoIP monitor
VoIP SIP and SKINNY quality analyzer and packet / audio recording tool
VoIPmonitor is open source network packet sniffer with commercial frontend for SIP SKINNY MGCP RTP and RTCP VoIP protocols running on linux. VoIPmonitor is designed to analyze quality of VoIP call based on network parameters - delay variation and packet loss according to ITU-T G.107 E-model which predicts quality on MOS scale. Calls with all relevant statistics are saved to MySQL or ODBC database. Optionally each call can be saved to pcap file with either only SIP / SKINNY protocol or... -
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BlackBelt Privacy Tor/i2p+WASTE+VidVoIP
Browse on Tor/i2p, Anon p2p Chat / FileTx, Conf / Video VoIP
Open Source - GPLv3 inc images. *** PLEASE NOTE: There are now 2 seperate versions here. *** One is Pre Firefox 57. The other is Post Firefox 57. *** For those providing mirrors, please enable your users to realize this. Vidalia Based, Tor as a Service Solution. MicroSip: enables FREE PC to PC video calling with no account sign-up and no middleman server. WASTE: enables FREE Conference VoIP, chat, file transfer and support. *** AI Powered *** Tor/i2p: enables safer... -
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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... -
Gemini 3 and 200+ AI Models on One PlatformBuild generative AI apps with Vertex AI. Switch between models without switching platforms.
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A project for a highly scalable protocol and framework and a collection of software tools for m2m (machine to machine communication) and IoT Internet of Things. The programs here works on Windows and Linux and are based around VSCP, The Very Simple Control Protocol. Repository is here: https://github.com/grodansparadis/vscp_software and here https://github.com/grodansparadis/vscp_firmware
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Vodia PBX
A Better PBX For Your Business
Vodia PBX has included a complete suite of business telephony features for on-premise and cloud-based telephony deployments. Sign up at https://vodia.com/en/register and receive your free Vodia license. -
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oreka
Enterprise telephony recording and retrieval system
Enterprise telephony recording and retrieval system with web based user interface. The project currently supports recording voice from VoIP SIP, Cisco Skinny (aka SCCP), raw RTP and audio sound device and runs on multiple operating systems and database systems. It can record audio from most PBX and telephony systems such as BroadWorks, Metaswitch, Asterisk, FreeSwitch, OpenSIPS, Avaya, Nortel, Mitel, Siemens, Cisco Call Manager, Cosmocom, NEC, etc... -
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Komendant ERP
ERP, IP PBX, Accounting, CRM, Time tracking, and etc.
Comprehensive business management system The full-cycle automation ERP system - [ Full Freeware ] IP PBX - Complete synchronization with CRM system - Call accounting for PBX (PBX billing) - Phone call control and monitoring - Phone recording systems Accounting, Financial management - Trade management, warehouse - Management and tax accounting Corporate information system - Client relations management system (CRM) - Project, tasks, orders management - On-line conferences... -
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SIPSL (SIP Service Layer) is a Programmable Session Border Controller (SBC) and SIP B2BUA, as defined in RFC3261. Written in c++ and multithreaded. The Application logic can be implemented by extending the application class and implementing the call backs.
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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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GetSmart is a download manager that supports Multi-Connection downloads. With Multi-Mirror search it can download different parts from these mirror servers simultaneously. It can 'act as proxy' to serve multiple clients. It can also works as a Daemon.
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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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VoIP Honey project provides a set of tools for building an entire honeynet, thus includes honeywall and honeypot emulating VoIP environments such as Asterisk PBX or OpenSer with fully configurable connections. Voip Honey runs on GNU/Linux and Windows Systems. It can be compiled for Mac OSX as well.
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nanoagi
Microscopic AGI framework.
Asterisk Gateway Interface often used when it is necessary to build complex IVR system. Nanoagi will help to do it. -
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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
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integral
Fast CGI-based AsteriskPBX cluster administration tool.
Allows OnLine monitoring of all kind Asterisk functioality: - Real-Time channel monitoring; - Queue Agents control; - Conference Administration; - Config edition; - Asterisk Core monotoring; -
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Voxy - the simplest way to integrate VoiceXML applications in Asterisk. Configure your dial plan with the URL of your VoiceXML application and it's done.
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skiputler makes your Topcom "Butler 4012 USB VoIP" phone work on Linux and Mac OS X. It's a userspace driver which connects the phone to Skype.
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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.
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Seagull is a multi-protocol traffic generator. Especially targeted towards IMS, Seagull supports Diameter (RFC3588 and all applications) over TCP/SCTP and IPv4/IPv6 , TCAP (over SS7 or Sigtran), XCAP over HTTP and Radius.
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ss7mon
ss7mon an SS7/SIGTRAN signalling monitor
ss7mon will implement the following features for SIGTRAN and SS7 family of protocols: - Message sniffer. - Call flow tracing. - Traces DB. - Distributed tracing. - QoS monitoring. - KPI monitoring. - Web interface. - CLI command line interface. - ...etc. -
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The ASN.1 Compiler
Go to github.com/vlm/asn1c for the latest version.
This ASN.1 compiler turns ASN.1 specifications into C code. The asn1c is shipped together with conformant BER/DER/XER/PER codecs. The X.509, GSM TAP3, MEGACO, RRC and LDAP encoding and decoding examples are part of the source code distribution. NOTE: THE asn1c PROJECT HAS LARGELY MOVED TO GITHUB: http://github.com/vlm/asn1c -
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An open source software project delivering converged voice and data applications. The StarPound platform enables you to rapidly develop voice and data solutions using the power of business process modeling. More info: http://www.starpound.org.
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Siemens EWSD PSTN Exchange can dump its configuration into regenfiles (or commandfile, cmdfile). rg2xml interprets these files and transforms information to XML
