Search Results for "sip server"
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Showing 67 open source projects for "sip server"
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Sippts
Set of tools to audit SIP based VoIP Systems
Sippts is a set of tools to audit VoIP servers and devices using SIP protocol. Sippts is programmed in Python and it allows us to check the security of a VoIP server using SIP protocol. You can freely use, modify and distribute. If modified, please put a reference to this site. Most security tools can be used for illegal purposes, but the purpose of this tool is to check the security of your own servers and not to use to do bad things. I am not responsible for the misuse of this tool. Sippts... -
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Routr
The future of programmable SIP servers
Lightweight sip proxy, location server, and registrar that provides a reliable and scalable SIP infrastructure for telephony carriers, communication service providers, and integrators. Routr provides all the tools required to deploy your VoIP network, including a command-line for remote server control. It can also be controlled via a RESTful API or a web-based GUI. Routr includes all tools for deploying your VoIP network. It offers remote server control via command-line, RESTful API, or a user... -
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VirtualBuddy
Virtualize macOS 12 and later on Apple Silicon
VirtualBuddy can virtualize macOS 12 and later on Apple Silicon, with the goal of offering features that are useful to developers who need to test their apps on multiple versions of macOS, especially betas. -
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*Until you install the operating system, the default user = live / no password. DragonOS FocalX (22.04) and DragonOS Focal (20.04) are out-of-the-box Lubuntu based x86_64 operating systems for anyone interested in software defined radios. All source installed software is located in the /usr/src directory while the remaining software was installed by package managers. What is DragonOS and why do you want it? The shortest distance between two points is a straight line. DragonOS...
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Our robust pre-built connectors and our no-code, drag-and-drop interface makes it easy and fast to automatically sync vendors, invoices, and invoice payment data between Tipalti and your ERP or accounting software.
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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 SIP/RTP... -
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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 browsing. Tor... -
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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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Cisco TMS PhoneBook - PHP Version
Emulates Cisco TMS PhoneBook for CMM or Endpoints with PHP
... new participants to a existing conference. The search is regular expression and the filter compare the user query with the endpoint name or H323/SIP dial string from the database. All you need is a Web server (tested with apache) with PHP and "RewriteEngine on" to make it work Optionally you can use a SQL server to have a real database, but I used a simple array as db. -
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OpenSIPS (former OpenSER) is an GPL implementation of a multi-functionality SIP Server that targets to deliver a high-level technical solution (performance, security and quality) to be used in professional SIP server platforms. IMPORTANT: this is no longer the main hosting for the project. This was moved on GITHUB - https://github.com/OpenSIPS/opensips
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Protocol plugin for Office 365/Lync/OCS
for Adium, Pidgin, Miranda and Telepathy IM Framework
Please make sure to read this page: http://sourceforge.net/p/sipe/wiki/Frequently%20Asked%20Questions/ A third-party Pidgin/Adium/Miranda/Telepathy plugin for the extended version of SIP/SIMPLE used by various products: * Skype for Business * Microsoft Office 365 * Microsoft Business Productivity Online Suite (BPOS) * Microsoft Lync Server * Microsoft Office Communications Server (OCS 2007/2007 R2) * Microsoft Live Communications Server (LCS 2003/2005) -
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Accent Tool Suite
Accent interfaces and controls systems through goals and policies
... and devices to support home care o Sensor Networks: sensor networks, particularly managing wind farms However, the approach is generic and extensible for managing other kinds of systems. Accent is essentially software. As most of Accent in in Java, it needs at least a Java Runtime Environment but more likely a Java Software Development Kit. To interact with the real world, Accent needs relevant hardware such as a communications server (softswitch) or home automation equipment. -
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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... It is amongst others being used in Call... -
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SIP_SERVER
SIP_SERVER_METI
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Andspoilt
Run interactive android exploits in linux.
Andspoilt is a command line user interface designed to easily exploit android devices. Run interactive android exploits in linux by giving the users easy interface to exploit android devices Uses an intergration with Metaspoilt Framework by giving the user an easy interface to create payloads and launch android exploits. Current Additional feature is a simple web server for file distribution. -
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TOFFEE-DataCenter
Linux Server and Data-Center WAN Optimization
TOFFEE-DataCenter is the new TOFFEE variant meant for Data-Center, Server, HPC, Load Balancing, Cluster/Cloud Computing and SDN deployments. With TOFFEE-DataCenter (and or TOFFEE), OEMs and other equipment vendors can make use of this platform/stack, integrate, port and build WAN Optimization devices/appliances on to their existing product range. Or build a dedicated WAN Optimization hardware product series via the same. -
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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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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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Mobicents is the leading Open Source VoIP Platform. It is the First and Only Open Source Certified implementation of JSLEE 1.1 (JSR 240), and SIP Servlets 1.1 (JSR 289). Mobicents also includes a powerful and extensible Media Server.
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Git repo: https://github.com/asipto/siremis Web management interface for Kamailio (OpenSER) - handle subscriber profiles, access control lists, accounting, least cost routing and load balancing, monitoring charts, xmlrpc communication with SIP server
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SIP Anonymization Tool (SiAnTo)
Small and effective program for SIP traces anonymization
The Session Initiation Protocol (SIP) is a signaling communications protocol widely used nowadays for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks. A good way to design optimization techniques for SIP deployment would be to analyze SIP traffic from existing networks. However, publicly available analyses of SIP traffic are rare and thus not a lot of knowledge exists about typical behavior of a SIP server (as opposed... -
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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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SSL Diagnos
SSL Strength Evaluation and Test Utility
SSL Diagnos is used to test SSL strength; get information about SSL protocols (pct, ssl2, ssl3, tls, dtls) and cipher suites. It can also be used for testing and rating ciphers on SSL clients. It has also specific support for pop3s, sip, smtp and explicit ftps. Tests for heartbleed (including dtls). Furthermore a separate tool, SSLPressure, not using openssl can be used to check the whole spectrum of possible SSL protocols on a server. Can also be used for testing ssl for mssql-servers... -
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KAMAILIO (OpenSER) - robust, secure and scalable Open Source (GPL) SIP (RFC3261) server implementation with large features set (over 90 extension modules). As of May 2009, source code is hosted by GIT repository at http://sip-router.org
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bv-tapi is a SIP TAPI bridge. It supports click to dial and screen popping. It is compatible with Freeswitch and babblevoice.com, but potentially should work with any SIP server which supports RFC 4235 (not Asterisk!). Source code now hosted at github. https://github.com/tinpotnick/bv-tapi.git
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CafeSip - Look what Java and SIP can do
A suite of open-source tools and frameworks for creating SIP apps
Session Initialtion Protocol (SIP) is widely used for telephone services over the Internet. CafeSip provides a suite of open-source tools and applications for creating customized SIP services and applications using the Java.