Search Results for "sofia-sip"
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91 projects for "sofia-sip" with 2 filters applied:
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Auth0 B2B Essentials: SSO, MFA, and RBAC Built InAuth0's B2B Essentials plan gives you everything you need to ship secure multi-tenant apps. Unlimited orgs, enterprise SSO, RBAC, audit log streaming, and higher auth and API limits included. Add on M2M tokens, enterprise MFA, or additional SSO connections as you scale.
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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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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. -
Streamline Azure Security with Palo Alto Networks VM-SeriesImprove your security posture and reduce incident response time. Use the VM-Series to natively analyze Azure traffic and dynamically drive policy updates based on workload changes.
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jfBroadcast
VoIP/SIP Autodialer
VoIP/SIP AutoDialer. Broadcasts a message with the option to transfer person to another number. Now includes new survey options. -
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pcapsipdump is libpcap-based SIP sniffer with per-call sorting capabilities. It writes SIP/RTP sessions to disk in a same format, as "tcpdump -w", but one file per SIP session (even if there is thousands of concurrent SIP sessions). Getting started: http://pcapsipdump.sf.net/
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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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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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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 Centers and Contact Centers for Quality monitoring (QM) purposes. -
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Taki
SIP softphone
Cross-platform SIP softphone -
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Callflow Sequence Diagram Generator
Callflow Sequence Diagram Generator
The callflow sequence diagram generator is a collection of awk and shell scripts that will take a packet capture file that can be read by wireshark and produce a time sequence diagram. This is useful to view & debug SIP callflows or other network traffic -
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wxCommunicator is a cross platform SIP softphone written in C++ utilizing customized sipXtapi user agent library and wxWidgets 2.8.9 GUI library. For a list of supported features see http://wxcommunicator.sourceforge.net/features.html .
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jpbxlite
Java VoIP/SIP PBX system (replaced by jfPBX)
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 -
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SIP Data Filter (SiDaFir)
Simple and efficient tool for SIP trace filtering
...Aim of this project is to offer a simple and effective, yet well-configurable, tool allowing for SIP trace filtering - the SIP Data Filter (SiDaFir). -
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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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baresip
Baresip is a modular SIP User-Agent with audio and video support
Baresip is a portable and modular SIP User-Agent with audio and video support. the latest source code can be found here: https://github.com/alfredh/baresip -
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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).
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Sippy B2BUA and RFC3261 SIP Stack
This project has been relocated to http://github.com/sippy/rtpproxy h
This project has been relocated to http://github.com/sippy/rtpproxy http://github.com/sippy/b2bua -
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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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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 to, for example, HTTP servers). ... -
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Artemisa is a honeypot for VoIP (SIP) networks. It is designed to connect to a VoIP domain as a user-agent backend in order to detect malicious activity at an early stage and also adjust the policies of the enterprise in real-time.
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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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LuaSofia is a Lua binding of Sofia-Sip library. Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification. We decided to use git, source code can be get at: git://github.com/ppizarro/luasofia.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.