Search Results for "sip"

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.

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.

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.

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.

Accent (Advanced Component Control Enhancing Network Technologies) is a comprehensive tool suite that interfaces a variety of communications systems and allows these systems to be controlled through goals (high-level user aims) and policies (lower-level system rules). Accent has been applied to the domains of: o Call Control: for telecommunications, particularly call control in Internet telephony o Home Care: for home automation and telecare, particularly for domestic appliances...

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 Centers and Contact Centers for Quality monitoring (QM) purposes.

...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. .

jPBXLite is a VoIP/SIP PBX. Supports SIP extensions, voicemail, trunks, conferences, queues (ACD) and an IVR system. Support video conferencing with jPhoneLite/1.4.0. NOTE:THIS PROJECT WAS RENAMED AND IS NOW jfPBX. Please go to jfpbx.sourceforge.net

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.

...This allows network engineers to quickly identify the behavior of their IMS network without tracing through log files or raw captures. Distributed SIPFlow captures and displays SIP and Diameter messages from remote hosts in an intuitive graphical interface. Each SIPFlow console interfaces with one or more SIPSniffers distributed across a SIP/IMS network and displays the SIP and Diameter call flows as reported by those sniffers. This is useful if your network of servers span multiple network segments or switches. ...

3M Standard Interchange Protocol (SIP) is an industry standard protocol by 3M to allow automatec check out terminals communicate with library systems. JSIP2 is an Open Source (MIT) library that implements the 3M SIP version 2 protocol including the 3M SIP2 Extensions for SIP2 clients. The library contains the necessary functionality for SIP2 client socket based implementation supporting all the SIP2 messages and fields, including also the fields defined in the the 3M SIP2 Extensions.

Peers is a very simple softphone. It's a SIP User-Agent, written in java, it works on windows, linux and mac. It can be used with SIP servers like opensips or asterisk IPBX. It supports G711 codec (PCMU and PCMA) and telephone-events (DTMF).

Client frontend built by Omnitor AB, Sweden. See README for instructions on how to build and run.

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.

IMS-Communicator is an IMS softphone based on the old version of the sip-communicator java project, implemented on top of the JAIN-SIP stack and the Java Media Framework API. It can be used with Open IMS Core (http://www.openimscore.org).

SERAdmin is a GUI interface between SIP Express Router (SER) and a SER administrator. This interface will extend the access of ser administration even from a remote location(through HTTP) and across Operating Systems. SERAdmin provides control over man

...It includes an MRCPv2 Server with ASR and TTS engines as well as an voiceXML interpreter so that you can deploy and run voiceXML applications. It integrates with VOIP PBX’s (like Asterisk) using SIP and RTP.

Siptest is a set of utilities for checking a computers access to a sip ip-telephony server thru firewalls and gateways.

SIP Inspector is a tool written in JAVA to simulate different SIP messages and scenarios. You can create your own SIP signaling scenarios, customize SIP messages and monitor incoming and outgoing messages. The tool can play RTP streams from a pcap fi

Open source implementation of the Open Mobile Alliance (OMA) Location in SIP (LOCSIP) requirements, providing extensions to the Session Initiation Protocol (SIP) for conveying locations requests and responses to enable location based services.

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

jSipgateConnect library allows Java-based access to all functions specified in the 'Sip Advanced Messaging, Usability, Routing and Accounting Interface' specification.

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.

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.

miTester for SIP is an automated SIP testing tool designed and developed to take care of the complex pre-deployment testing of SIP applications easily. This SIP testing tool can be used to simulate SIP call-flows & automate functional, regression tests.