Search Results for "sofia-sip" - Page 7
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Showing 323 open source projects for "sofia-sip"
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The main aim of the project is to built a application on the top of IMS which will convert text to speech. If one user sends text from one side other user will receive it as Voice. This application can be use as simple Voice SMS service
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Máquina virtual com instalação básica do asterisk no centos para fins de testes. Inicie a máquina e configure um sip-phone com os logins e senhas contidos em /etc/asterisk/sip.conf.
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Txt2XMLParser is a versatile and generic text to xml converter library. Based on Regular Expression it can be configured using a single xml file allowing to parse any txt file format (csv , generic txt files, protocol formats such as SIP)
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OpenBTS
GSM L1-L3 stack with SIP network interfaces
OpenBTS is an implementation of the GSM air interface (Um) that allows cellular handsets to be used directly as SIP endpoints. It uses a software-defined radio to generate its air interface and uses Asterisk or yate as its network interface. -
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This projects is a simple Media Gateway, build under Sofia-Sip and Gstreamer libraries. It was build for a project that uses a MediaGateway and Open-IMS to distributed audio publicity.
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Fluent is a suite of tools for the management of diverse VoIP networks. Fluent RTE, a reporting and templating engine, is at the core. Add-on modules are available to manage provisioning, CDR, feature sets, routing and more.
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AuDA (Availability under DoS Attacks) security testing environment. Doing automated & reuseable tests. First target it is written for is SIP system. But it is easily extensable and may adopted to other internet services.
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Simu server project is aimed for IP telephony (VoIP) between classical SIP phones and Flash streaming applications.
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Dubtel One is a Cloud API for quickly building advanced voice and chat applications. The API was developed using industry standards including XML, SIP and XMPP.
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SIP.NET is SIP stack .NET library written in C#. SIP message parser realized on Deterministic Finite-state Machine (DFA parse all SIP message for one pass). DFA generates automatically from Augmented Backus–Naur Form from RFC3261.
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VitoshaTrade is Distributed Artificial Neural Network trained by Differential Evolution for prediction of Forex. Project development is in Sofia, Bulgaria. Vitosha is a mountain massif, on the outskirts of Sofia, the capital of Bulgaria.
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Live-cd com Servidor SIP baseado em opensips. O Servidor de Telefone Livre transformará um simples PC numa central SIP multidominio. Estamos construindo a documentacao do projeto na wiki do sistema trac.
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jSipgateConnect library allows Java-based access to all functions specified in the 'Sip Advanced Messaging, Usability, Routing and Accounting Interface' specification.
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This project is a SIP based softphone developed as a Java applet that can be embedded into a website. The applet has a simple user interface and is intended to be used by journalists to make and deposit audio recordings via a SIP proxy.
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Rabboni VOIP is an Open Source Cross Platform Softphone developed with trolltech QT LIBRARY it uses PJSIP - Open Source SIP Stack
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OS SIP applet can be used to make a VoIP call from the web page. Integration can be made using java script API, which is provided by the SIP applet.
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Telepathy-SofiaSIP is a SIP connection manager (protocol plugin) for the Telepathy framework (http://telepathy.freedesktop.org). It's based on the Sofia-SIP stack.
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SerMyAdmin
Web interface for OpenSIPS
... would like to say thanks to all the people that in some way contributed with the project. The svn code is available to anyone who wants to maintain the project. Most of our development efforts are now focused in a new commercial product based on OpenSIPS called SIP Pulse, completely rewritten in Java and Glassfish www.sippulse.com supporting pre and post paid users with billing and reseller portal. -
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Open Unified Recording (OUR) is a full featured Linux based Open Source VoIP/SIP Call Recording engine, indexing and retrieval system. The system resides on the network and passively captures SIP sessions. Project is sponsored by UnifiedRecording.com
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The program captures SIP traffic (from port-mirroring on a switch) and provides statistics about the SIP message latency. The latency is the delta between incoming and outgoing messages to a SBC / proxy that process SIP messages.
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A simple lightweight instant messaging and VoIP client wirtten in Java. It implements the SIP protocol as described in RFC 3621 and other related RFCs and implementation requierements.