Search Results for "linux pxe server"
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Showing 40 open source projects for "linux pxe server"
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Go From AI Idea to AI App FastAccess Gemini 3 and 200+ models. Build chatbots, agents, or custom models with built-in monitoring and scaling.
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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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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... -
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BlackBelt Privacy Tor/i2p+WASTE+VidVoIP
Browse on Tor/i2p, Anon p2p Chat / FileTx, Conf / Video VoIP
...A usability enhanced Privacy Pack. Use alongside Tor Browser Bundle. An installer, for : Vista 32/64, Win7 32/64, Win8 32/64, Win10 32/64, Win11, Linux (Wine) -
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Stuntman - STUN server and client
High performance, production quality STUN server and client library
New version 1.2. This is the code to STUNTMAN - an open source STUN server and client code by john selbie. Compliant with the latest RFCs including 5389, 5769, and 5780. Also includes backwards compatibility for RFC 3489. ICE and WebRTC ready. Version 1.2 compiles on Linux, MacOS, BSD, and Solaris. Supports the STUN protocol on both UDP and TCP for both IPv4 and IPv6. Windows binaries are also provided. -
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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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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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GetSmart is a download manager that supports Multi-Connection downloads. With Multi-Mirror search it can download different parts from these mirror servers simultaneously. It can 'act as proxy' to serve multiple clients. It can also works as a Daemon.
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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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This project implements a simple STUN server and client on Windows, Linux, and Solaris. The STUN protocol (Simple Traversal of UDP through NATs) is described in the IETF RFC 3489, available at http://www.ietf.org/rfc/rfc3489.txt
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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... -
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Seagull is a multi-protocol traffic generator. Especially targeted towards IMS, Seagull supports Diameter (RFC3588 and all applications) over TCP/SCTP and IPv4/IPv6 , TCAP (over SS7 or Sigtran), XCAP over HTTP and Radius.
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An open source software project delivering converged voice and data applications. The StarPound platform enables you to rapidly develop voice and data solutions using the power of business process modeling. More info: http://www.starpound.org.
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SEMS is a GPLv2+ media and application server for SIP based VoIP services. Basic applications like announcement, pre-call announcement, RBT, conference, voicemail, mailbox etc. and lots of example applications available.
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Simu server project is aimed for IP telephony (VoIP) between classical SIP phones and Flash streaming applications.
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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.
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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.
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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).
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This project has been superseded by OpalVoip (https://sourceforge.net/projects/opalvoip/) and H323Plus (https://sourceforge.net/projects/h323plus/) The OpenH323 project provides full featured, interoperable, Open Source implementation of the ITU H.323 teleconferencing protocol that can be used by personal developers and commercial users without charge.
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Bluetooth Configurable Remote Control Server This application executes commands asociates to messages strings.The messages can be send by a RFCOMM bluetooth connection.A sample cliente Bluetooth Configurable Remote Control Client(BCRCC) is also provided.
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IM,WebIM and Sip voip tool. Client is Win32, Server is CentOS 5. support video,audio, file trans, sip call.
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Ernie integrates Web 2.0 design principals with next generation communications technologies, using VoIP, presence and Python. Ernie is used wherever a real time communications enabled web page or application is hosted on a server.
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Asterisk Manage API and FAGI C++ Framework. Implemented with high efficient OS independent Sockets wrapper (ACE). Provide C++ API wrappers for Asterisk Manager API and FAGI. A Manage API relay(proxy) server and a FAGI socket server will be implemented.
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VVOIP is voice and video over IP is a P2P application without any protocol like SIP (no server is required) etc.. application. It use decent codecs for video and voice encrypted streams over IP. Moreover it is also comptiable with VoIP modems, you'd con
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Bayonne is supported from Savanna.gnu.org. Please visit there. Sourceforge is generally depreciated except for mailing lists, but updated releases will be posted here from time to time.
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UCE is H323 Telephony. UCE includes whole features of GSM phones and works with different architectures like PSTN. UCE provides voice commands,make it easy for blind users with sound server support.UCE made in TURKEY (Uraz-Cihad-Ersin) Atilim University
