Search Results for "sip"
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Showing 221 open source projects for "sip"
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Sip
Sip: A MUD Client!
•ANSI-TRUE: All the colors you could hope for. •MXP: Pictures, sounds, clickable text, and mini-frames. •MSP: Sound and Music •GMCP: Access server information out-of-band + Client.Login, Client.Media. •MSDP: Access server information out-of-band •Aliases: Customize your commands, or trigger a myriad of events. •Triggers: Cause a custom action when the mud sends a particular message. •Scripts (JavaScript): Go wild with customizing your mud client. •Timers: Cause some action to occur... -
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Sippts
Set of tools to audit SIP based VoIP Systems
...I am not responsible for the misuse of this tool. Sippts is a set of tools to audit VoIP servers and devices using SIP protocol. Sipscan is a fast scanner for SIP services that uses multithread. Sipscan can check several IPs and port ranges and it can work over UDP or TCP. Sipexten identifies extensions on a SIP server. Also tells you if the extension line requires authentication or not. Sipexten can check several IPs and port ranges. -
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SIPVicious
Security tools that can be used to audit SIP based VoIP systems
SIPVicious OSS has been around since 2007 and is actively updated to help security teams, QA and developers test SIP-based VoIP systems and applications. Open-source security suite for auditing SIP based VoIP systems. Also known as friendly-scanner, it is freely available to help pentesters, security teams and developers quickly test their SIP systems. Download the latest source code from git or the latest release, send pull requests and open issues. -
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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. -
MongoDB Atlas runs apps anywhereMongoDB Atlas gives you the freedom to build and run modern applications anywhere—across AWS, Azure, and Google Cloud. With global availability in over 115 regions, Atlas lets you deploy close to your users, meet compliance needs, and scale with confidence across any geography.
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Ovmeet
Video conferencing and collaboration platform
...Supports WebRTC, RTMP, RTSP, SIP protocols, enabling interoperability with streaming devices, hardware endpoints, and different media sources. -
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Android Contacts, Reborn
Android Contacts API Library written in Kotlin
This library provides a complete set of APIs to do everything you need with Contacts in Android. You no longer have to deal with the Contacts Provider, database operations, and cursors. Whether you just need to get all or some Contacts for a small part of your app (written in Kotlin or Java), or you are looking to create your own full-fledged Contacts app with the same capabilities as the AOSP Android Contacts app and Google Contacts app, this library is for you. -
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wxPython Project Phoenix
wxPython's Project Phoenix. A new implementation of wxPython
wxPython, the cross-platform GUI toolkit for the Python language. With wxPython software developers can create truly native user interfaces for their Python applications, that run with little or no modifications on Windows, Macs and Linux or other Unix-like systems. Welcome to wxPython's Project Phoenix! Phoenix is the improved next-generation wxPython, "better, stronger, faster than he was before." This new implementation is focused on improving speed, maintainability and extensibility.... -
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JavaCPP
The missing bridge between Java and native C++
JavaCPP provides efficient access to native C++ inside Java, not unlike the way some C/C++ compilers interact with assembly language. No need to invent new languages such as with SWIG, SIP, C++/CLI, Cython, or RPython. Instead, similar to what cppyy strives to do for Python, it exploits the syntactic and semantic similarities between Java and C++. Under the hood, it uses JNI, so it works with all implementations of Java SE, in addition to Android, Avian, and RoboVM (instructions). More specifically, when compared to the approaches above or elsewhere (CableSwig, JNIGeneratorApp, cxxwrap, JNIWrapper, Platform Invoke, GlueGen, LWJGL Generator, JNIDirect, ctypes, JNA, JNIEasy, JniMarshall, JNative, J/Invoke, HawtJNI, JNR, BridJ, CFFI, fficxx, CppSharp, cgo, pybind11, rust-bindgen, Panama Native, etc.), it maps naturally and efficiently many common features afforded by the C++ language and often considered problematic, including overloaded operators, class and function templates, etc. -
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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/RTCP/T.38/udptl protocols. ... -
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Siproxd is a proxy/masquerading daemon for the SIP protocol. It allows SIP clients (softphones & hardphones) to work behind an IP masquerading firewall or router.
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IssabelPBX
Issabel PBX - Unified Communications
Open Source and Unified Communications partners created a new platform based on an Elastix® fork (currently purchased by 3CX) to provide the community with continuity, peace of mind and support needed to continue with their PBX and operation developments. Contribute to the funding of Issabel on https://www.patreon.com/issabel -
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jfPhone
VoIP/SIP SoftPhone (formerly known as jphonelite)
jfPhone is a VoIP/SIP SoftPhone for Desktops. -
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ictpbx-community-edition
Open source multi-tenant PBX built on ICTCore and FreeSWITCH
...Tenants are separated properly rather than bolted on, so each one gets its own extensions, DIDs, trunks and routing without seeing anyone else's. The usual PBX building blocks are there: extensions and device provisioning, inbound and outbound routing, SIP trunks, IVR, ring groups, queues and voicemail, all administered through the web interface rather than by editing dialplan files. Released under MPL-2.0. There is an Ansible role for installing it, and the web interface is a separate repository if you want to build on it. -
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ictcore
Unified communications framework and REST API for developers
ICTCore is the communications framework the rest of our open source stack is built on. It sits above FreeSWITCH and gives web developers a REST API for voice calls, fax, SMS and email instead of a dialplan to learn. The idea is that a communications workflow should be something you write in your own application. You describe what should happen, and ICTCore deals with the switch, the media and the state machine underneath. It is the engine behind ICTFax and the open source ICTDialer, so... -
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008
Open-source event-driven AI powered Softphone
...This agent utilizes embedded artificial intelligence models operating directly on the softphone, ensuring efficiency and reduced operational costs. This project is a WebRTC softphone, and communication is achieved via SIP over a socket. Leading PBX systems like Asterisk or Freeswitch support socket connections. If your provider does not offer this feature, consider using a SIP proxy such as Kamailio, Opensip or Routr. The softphone is internally configured using a JSON definition. The configuration file can be loaded from either a server or a local file. 008 reads the file only once. -
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Fishjam
General purpose media server. Supports WebRTC, HLS, RTSP, SIP
Fishjam is a real-time multimedia streaming toolkit for building low-latency video and audio applications. The GitHub repository refers to the original open-source Fishjam media server, which was a general-purpose media server written in Elixir and is no longer actively maintained. The broader Fishjam project has since evolved into a platform for WebRTC-based live video, audio, and messaging infrastructure. It is designed for use cases such as video conferencing, live streaming, classrooms,... -
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SIP.js
A simple, intuitive, and powerful JavaScript signaling library
A JavaScript library that enables WebRTC-based real-time communication over the SIP (Session Initiation Protocol), allowing voice and video calling in web applications. -
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Slipstream
NAT Slipstreaming allows an attacker to remotely access any TCP/UDP
...The attack is able to bypass browser port restrictions by fragmenting or massaging packets so that the “exploit payload” lands in a packet boundary that gets parsed by the NAT/ALG as a legitimate protocol packet (e.g. SIP or H.323). Once successful, the NAT/firewall is deceived into opening a “hole” (port forwarding) back to the internal host, enabling the attacker to connect directly to internal services. -
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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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VitalPBX
Unified Communications System - PBX System
VitalPBX is a free telephone and communications PBX system for companies. It is a complete platform that can be installed on the physical hardware on the site or as a hosted application. VitalPBX acts as the upper layer interface for the Linux base and then Asterisk (one of the most popular communication toolkits in the world). For this reason, VitalPBX is the graphic user interface between you and the complex world of modern communications. VitalPBX will help you implement a secure... -
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Cisco TMS PhoneBook - PHP Version
Emulates Cisco TMS PhoneBook for CMM or Endpoints with PHP
...Anyway, this tool emulates the TMS PhoneBook and allows you to search and add 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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Calis-p
Estimates delta13C of species in a microbiome from proteome data
Calis-p (The CALgary approach to ISotopes in proteomics) is a java application to estimate isotopic composition (e.g. delta13C or delta15N) of individual species in a microbial community from a proteomic dataset. Calis-p 2.0 handles both natural isotope abundances and data from labelling experiments such as stable isotope probing (SIP). It requires a mzIdent (or target spectrum match) and mzML files as the input and requires about 1 min per mzML file with 10 threads and needs <10 Gb of RAM. It has been tested with data from various nano liquid chromatography/Orbitrap platforms. For successful SIP, it is extremely sensitive, but requires the 13C fraction to remain below 10%. ... -
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Jitsi Desktop
Jitsi is an audio/video and chat communicator that supports protocols
Jitsi Desktop is a free open-source audio/video and chat communicator that supports protocols such as SIP, XMPP/Jabber, IRC and many other useful features. Jitsi Desktop is the heritage of Jitsi Meet. While some components are still used in e.g. Jigasi, the project is not actively developed anymore. Improvements, bugfixes and builds are entirely based on community contributions. -
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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.
