Search Results for "sofia-sip" - Page 2
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Showing 56 open source projects for "sofia-sip"
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Stop Storing Third-Party Tokens in Your DatabaseRolling your own OAuth token storage can be a security liability. Token Vault securely stores access and refresh tokens from federated providers and handles exchange and renewal automatically. Connected accounts, refresh exchange, and privileged worker flows included.
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Enterprise-grade ITSM, for every businessFreshservice is an intuitive, AI-powered platform that helps IT, operations, and business teams deliver exceptional service without the usual complexity. Automate repetitive tasks, resolve issues faster, and provide seamless support across the organization. From managing incidents and assets to driving smarter decisions, Freshservice makes it easy to stay efficient and scale with confidence.
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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.
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Sipana is a distributed SIP monitoring tool to monitor the SIP signaling behavior using sequence diagrams and providing SIP end-to-end performance metrics through a centralized Web interface. For more information please visit http://sipana.org/
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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).
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Build Agents and Models on One PlatformGemini Enterprise Agent Platform is Google Cloud's comprehensive platform for developers to build, scale, govern, and optimize agents and models. Choose from Google's most advanced models and third-party models like Anthropic's Claude Model Family.
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Among the functionality: Include a large variety of codecs (G711, GSM, and SPEEX) - Protocol SIP - Other technical functionalities the support of DTMF (tonalities) although support ENUM (to employ numbers of SIP instead of the addresses of SIP).
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With SIP Proxy you will have the opportunity to eavesdrop and manipulate SIP traffic. Furthermore, predefined security test cases can be executed to find weak spots in VoIP devices. Security analysts can add and execute custom test cases.
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HATEFNA هاتفنا .....http://www.hatefna.org ..... (PC&Mobile)IP SoftPhone over SIP and ASTERISK using JAVA for PC and Symbian C++ for Symbian Mobiles Phones ..... برمجيات للمهاتفة ببروتوكول الإنترنت
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Softphone SIP com voz, vídeo e Mensagem Instantânea. DTMF, Nat-Traversal, Suporte a Skin(Macromedia Flash), SMS, Historico de Chamadas, Presença, Hold, Auto-Answer e etc. Sip Softphone with Voice, Video and IM.
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CPLed is an OpenSIPS tool for editing CPL scripts in a friendly and easy graphical way. It can be used as a standalone application or embedded in a web page as applet. It also provide CPL script transport functionalities via SIP and HTTP protocols.
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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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Open Source IM and voice client using Jabber and SIP protocols, with great audio quality thanks to speex and a nice and clean interface. Programmed in Java and C++. It supports chat, voice and file transfers.
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This project is an implementation of the Sip Back-to-Back User Agent. The B2BUA acts as a user agent to both ends of a Session Initiation Protocol (SIP) call.
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SIP SoftPhone desenvolvido em Java baseado na JAIN-SIP, JMF e Sip-Communicator 1.0. Possui também o envio de SMS, Agenda, NAT Traversal, DTMF Send/Receive, CallTo e etc. A Java SIP SoftPhone based in JAIN-SIP, JMF and Sip-Comminicator 1.0.
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'The Genie' is a Java based SIP User Agent that can talk to any other SIP User Agent via SIP Proxy or Asterisk. Also Planned: 1-click sign-in to Yahoo or any other site by storing hashes of user/passwd pairs in a 'Wallet' and they can be accessed onl
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UChat is a JAVA and JPF (Java Plugin Framework) based application, that aims to provide a modular structure for chat APIs. SIP is mostly completly integrated (JAIN-SIP API from NIST), and functional (text and audio chat). Jabber (Muse API) is on the way
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The purpose of this project is to write a plugin for the Openexchange (OX) groupware server (http://www.openexchange.com) that supports various VoIP (SIP) functionality, like direct dialing a SIP telephone, broadcasting, messaging, etc.
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Universal Transport For Sip, Including UDP,TCP,HTTP Proxy Tunnelling,Sock4/5,STUN,UPNP,TURN, also a extension in sip for net tunnelling establish.
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JOpenPhone is an opensource UA (User Agent) JAVA application that allows users to connect to a voip servers using various voip protocols such as SIP, H323.
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A pure java implemtation of the Session Initation Protocol (RFC 3261). SIP is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants.
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PNX system develops the middle-ware source code to glue Asterisk with a number of powerful telephony products such as: 1) OpenH323 H.323 stack 2) Vovida SIP stack 3) Bayonne Voice Automation Platform The advantages? An advanced IP PBX supporting the wide
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SipSpy is a distributed monitoring tool for SIP networks. SpyAgents run on each of the nodes to be monitored, and a SipSpy connects to each of these nodes, receiving information and displaying it in real-time for all the SIP packets monitored.
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sippet
C++ SIP stack based on Chrome source code
This C++ library has been designed as a Chrome SIP stack. Sippet is an open-source SIP User-Agent library, compliant with the IETF RFC 3261 specification. It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and P2P communication services. The main target was to enable Javascript applications to use UDP, TCP and TLS transports along WebSocket. -
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The G.O.N.E. is a softphone (or soft phone) running over the web, fully multi-plataform, it implements the SIP protocol, and is built to work on any SIP server, like Asterisk, and others. GONE will work on a complex sistem, but this will be showed a bit
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