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Creacode SIP Application Server Real-time SIP call controller and IVR product for carrier-class VoIP networks; Dialogic Corporation Powermedia Media Servers, audio and video SIP IVR, media and conferencing servers for Enterprise and Carriers. Dialexia VoIP Softswitches, IP PBX for medium and enterprise organizations, billing servers.
Kamailio, formerly OpenSER (and sharing some common history with SIP Express Router (SER)), is an SIP server licensed under the GPL-2.0-or-later license. It can be configured to act as a SIP registrar, proxy or redirect server, and features presence support, RADIUS / syslog accounting and authorization, XML-RPC and JSON-RPC-based remote control, SQL and NoSQL backends, IMS / VoLTE extensions ...
SipXecs is a free software enterprise communications system. [1] It was initially developed by Pingtel Corporation in 2003 as a voice over IP telephony server located in Boston, MA. [2] The server was later extended with additional collaboration capabilities as part of the SIPfoundry project.
FreeSWITCH is a free and open-source telephony software for real-time communication protocols using audio, video, text and other forms of media. The software has applications in WebRTC, voice over Internet Protocol (VoIP), video transcoding, Multipoint Control Unit (MCU) functionality and supports Session Initiation Protocol (SIP) features.
SIP Witch is released as free software under the terms of version 3 or later of the GNU General Public License (GPL). It is designed for Linux, macOS, BSD and Windows and planned support for Android. [1] In the popular Linux distributions Ubuntu and Fedora it may be installed directly from the standard package sources. [2] [3]
Jitsi (from Bulgarian: жици — "wires") is a collection of free and open-source multiplatform voice (VoIP), video conferencing and instant messaging applications for the Web platform, Windows, Linux, macOS, iOS and Android.
Linphone (contraction of Linux phone) is a free voice over IP softphone, SIP client and service. It may be used for audio and video direct calls and calls through any VoIP softswitch or IP-PBX. Linphone also provides the possibility to exchange instant messages.
JsSIP uses the SIP over WebSocket transport for sending and receiving SIP requests and responses, and thus, it requires a SIP proxy/server with WebSocket support. Currently the following SIP servers have been tested and are using JsSIP as the basis for their WebRTC Gateway functionality: