Search Results for "sofia-sip" - Page 4

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.

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.

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.

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.

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

Rabboni VOIP is an Open Source Cross Platform Softphone developed with trolltech QT LIBRARY it uses PJSIP - Open Source SIP Stack

PJSGUA is an Open Source SIP UAC based on PJSUA reference UAC from PJSIP project

... 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.

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.

This project provides a meter of QoS on VoIP using the protocol SIP Media Loopback

This is a SIP signaling layer to create a fully operative multipoint (video) conference server using SIP clients and RTP media streams in combination with strManager as a media management layer.

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.

A plugin for Windows ALG Proxy Service to modify SIP/RTP packets. Currently it doesn't work and needs other developers to help.

An XCAP server is used by XCAP clients to store data like buddy lists and presence policy in combination with a SIP Presence server that supports PUBLISH, SUBSCRIBE and NOTIFY methods to provide a complete SIP SIMPLE server solution.

Java SIP stack, originally based on mjsip.

Quality plugin for PJSGUA SIP client.

A high-performance implementation of a SIP Stack in Java.

A Sip live audio feeding agent. The agent captures audio from sound card and sends live audio stream(uLaw) to caller(sip phone) using RTP. It is based on Peers 0.3(http://peers.sourceforge.net/). Can be used in IP telephony to broadcast live audio.

Prototype testbed implementation of the IETF Media Server Control (MEDIACTRL) SIP Control Framework, comprehensive of both control and processing functionality (as in IMS MRF, Media Resource Function).

This project develops video conferencing and VoIP applications for Mac OS X. XMeeting is based on the work of the OpenH323 project which provides the libraries for H.323 and SIP support.

VoIPER is a VoIP security testing toolkit incorporating several VoIP fuzzers and auxilliary tools to assist the auditor. It can currently generate over 200,000 SIP tests and H.323/IAX modules are in development. It's also a damn cool project name ;)

FreeRADIUS module providing connectivity to CDRTool prepaid engine. Provides prepaid authentication for calls proxied by OpenSER and returns to OpenSER MaxCallDuration and user Credit. Can be easily extended to support other SIP or H323 devices.

SIP to Skype Gateway/Bridge/Converter/Adapter. Make and receive SIP to Skype Calls and Skype to SIP Calls. Call Skype users using speed dial or use SkypeOut. Make SIP calls from Skype using a SIP provider or SIP PBX. Use as a Skype Trunk with a PBX.

VoIP Toolkit / Call Control with Integrated Media. High-level Java API for creating SIP enabled VoIP applications. Suitable for either the desktop (softphone, phone applet, incoming call gatekeeper) or server-side (auto attendant, ACD, voicemail).

This project contains a SIP stack and server applications built on top of the stack, examples of which are: Stateless Proxy, Registrar,NAT Keepalives,MWI Notifier server. The server applications have been designed to work in association with Asterisk.