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.
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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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this project will be based on linux OS, will be used to collect SIP Metrics report sent from EneUsers, there will be a metrics report schema.and all data
will store in Database Mysql
the project is in planning
SkypeAsterisk is a full-featured PBX for Skype, SIP, IAX and traditional phone. It is suitable for connecting calls between Skype, PSTN, and IP-PBX. Features include voice mail, conference, call recording and call transferring. An bill system is under de
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
This project implements series of homework assignment from Columbia university course.
The main target is to create a SIP enabled thin audio client for Linux.
SpITAssassin will be an enhancement to the successful SpamAssassin to detect Spam over Internet Telephony. This projects extends the SpamAssassin-functionality to insert SIP-compatible headers marking possible Spit-calls.
The SIP2IMS gateway aims to enable simple SIP clients to access IMS networks and provide support for IMS features such as Service-Routes, AKA authentication, IP-Sec/TLS etc.
A Simple Middlebox Configuration Protocol (RFC 4540) implementation for Linux.
SIMCO is a signaling protocol that can be used by applications (such as SIP B2BUAs) to dynamically control firewalls and Network Address Translators (NATs).
Tools and libraries for dialing URI addresses directly from telephone's numeric keypad w/o phone directories.
Encoded uri (for instance, SIP VoIP phone number) consists of decimal numbers only and can be entered from any telephone or other device.
...This call is established by the vTiger user of the vTiger web based GUI. FreeSWITCH can be configured as a similar soft PBX and so the goal is to get vTiger to cause Freeswitch to establish SIP connections from the vTiger user's phone to the customer phone.
Open Source SIP Back to Back User Agent) and Billing system. Has an Embeded TCL interpreter making session control and session mediation limitless in flexibility. Useful for: Routing Engine, Calling Cards, SBC, Peering, Wholesale and more!