Search Results for "sofia-sip"
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Showing 55 open source projects for "sofia-sip"
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Ship Agents FasterGemini Enterprise Agent Platform lets you rapidly build, scale, govern and optimize production-ready agents grounded in your organization's data. The platform enables developers to build custom or pre-built agents for virtually any use case. New customers get $300 in free credits.
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Full-stack observability with actually useful AI | Grafana CloudBuilt on open standards like Prometheus and OpenTelemetry, Grafana Cloud includes Kubernetes Monitoring, Application Observability, Incident Response, plus the AI-powered Grafana Assistant. Get started with our generous free tier today.
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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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BlackBelt Privacy- Tor i2p WASTE VidVoIP
Browse on Tor / i2p - Anon p2p Chat / FileTx, Conf / Video VoIP
Open Source - GPLv3 inc images. *** PLEASE NOTE: There are now 2 seperate versions here. *** One is Pre Firefox 57. The other is Post Firefox 57. *** For those providing mirrors, please enable your users to realize this. Vidalia Based, Tor as a Service Solution. MicroSip: enables FREE PC to PC video calling with no account sign-up and no middleman server. WASTE: enables FREE Conference VoIP, chat, file transfer and support. *** AI Powered *** Tor/i2p: enables safer... -
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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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Vodia PBX
A Better PBX For Your Business
Vodia PBX has included a complete suite of business telephony features for on-premise and cloud-based telephony deployments. Sign up at https://vodia.com/en/register and receive your free Vodia license. -
Our Free Plans just got better! | Auth0You asked, we delivered! Auth0 is excited to expand our Free and Paid plans to include more options so you can focus on building, deploying, and scaling applications without having to worry about your security. Auth0 now, thank yourself later.
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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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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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Taki
SIP softphone
Cross-platform SIP softphone -
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MCU Media Server
SIP Video Multiconference Media Server with WebRTC support.
...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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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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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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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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A TAPI driver for SIP. SIPTAPI gives you a click2dial feature with any TAPI enabled application (e.g. MS Outlook) and SIP PBX/proxy. This is the "free" version of SIPTAPI. There is also an enhanced commercial version available at www.ipcom.at.
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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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DragonEyes
An application for multiparty video/audio conference
The DragonEyes is a linux software which enables multimedia, multiparty collaboration in applications such as group conferencing, distance learning, training and video telephony. All written in C++ and with a GUI frontend. The interface is based on Qt. The media backend is based on the GStreamer framework. Communication protocols include SIP, RTP. -
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PocketSIPMsg is a SIP instant messaging client for small devices that use Windows CE. The IM protocol is based on SIMPLE(IETF) working group's specification. The project is developed with eMbedded Visual C++ as "Dialog based application".
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Konference is (or better: will be) a video-conferencing application for KDE. Since the rewrite (2005/01/25) it supports SIP as the signalling protocol. No longer H323 folks.
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What is t38modem? From your application view point it's a fax/voice modem pool. From IP network view point it's a H.323/SIP endpoint with T.38 fax support. From your view point it's a gateway between an application and IP network. Works with HylaFAX.
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This is a open source media gateway controller. The protocol used is MEGACO/H.248. In a later version also SIP will be supported.
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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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VitoshaTrade is Distributed Artificial Neural Network trained by Differential Evolution for prediction of Forex. Project development is in Sofia, Bulgaria. Vitosha is a mountain massif, on the outskirts of Sofia, the capital of Bulgaria.
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OpenBTS
GSM L1-L3 stack with SIP network interfaces
OpenBTS is an implementation of the GSM air interface (Um) that allows cellular handsets to be used directly as SIP endpoints. It uses a software-defined radio to generate its air interface and uses Asterisk or yate as its network interface. -
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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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Open Unified Recording (OUR) is a full featured Linux based Open Source VoIP/SIP Call Recording engine, indexing and retrieval system. The system resides on the network and passively captures SIP sessions. Project is sponsored by UnifiedRecording.com
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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.