Deploy in 115+ regions with the modern database for every enterprise.
MongoDB 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.
Start Free
Build Securely on Azure with Proven Frameworks
Lay a foundation for success with Tested Reference Architectures developed by Fortinet’s experts. Learn more in this white paper.
Moving to the cloud brings new challenges. How can you manage a larger attack surface while ensuring great network performance? Turn to Fortinet’s Tested Reference Architectures, blueprints for designing and securing cloud environments built by cybersecurity experts. Learn more and explore use cases in this white paper.
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).
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.
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
Start building on Google Cloud with $300 in free credits. No commitment, no credit card required until you're ready to scale.
Launch your next project with $300 in free Google Cloud credits—no strings attached. Test, build, and deploy without risk. Use your credits across the entire Google Cloud platform to find what works best for your needs. After your credits are used, continue with always-free tier services. Only pay when you're ready to scale. Sign up in minutes and start exploring.
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.
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.
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.
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.
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.
Everything you need to build production-ready agents and models. Access 200+ Google and third-party AI models and tools.
Gemini 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.
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
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.
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.
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
Present is an attendance dialer for educational organizations. It currently will integrate directly into Powerschool SIS, with the potential to connect to other SIS systems as well. It can be used to make automated attendance and tardy calls