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Alcatel-Lucent 5060 IP Call server; Aricent SIP UA stack, B2BUA, proxy, VoLTE/RCS Client; AskoziaPBX; Avaya Application Server 5300 (AS5300), JITC certified ASSIP VoIP; Bicom Systems IP PBX for telecoms; Brekeke PBX, SIP PBX for service providers and enterprises; Cisco SIP Proxy Server, Cisco unified border element (CUBE), Cisco Unified ...
SipXecs is designed as a software-only, distributed cloud application.It runs on the Linux operating system CentOS or RHEL on either virtualized or physical servers. A minimum configuration allows running all of the sipXecs components on a single server, including database, all available services, and the sipXecs management.
SIP trunking is a voice over Internet Protocol (VoIP) technology and streaming media service based on the Session Initiation Protocol (SIP) by which Internet telephony service providers (ITSPs) deliver telephone services and unified communications to customers equipped with SIP-based private branch exchange (IP-PBX) and unified communications facilities. [1]
Asterisk is a software implementation of a private branch exchange (PBX). In conjunction with suitable telephony hardware interfaces and network applications, Asterisk is used to establish and control telephone calls between telecommunication endpoints such as customary telephone sets, destinations on the public switched telephone network (PSTN) and devices or services on voice over Internet ...
Elastix is a unified communications server software that brings together IP PBX, email, IM, faxing and collaboration functionality. It has a Web interface and includes capabilities such as a call center software with predictive dialing. The Elastix 2.5 functionality is based on open source projects including Asterisk, FreePBX, HylaFAX, Openfire ...
3CX, Inc., is a software development company and developer of the 3CX Phone System. The 3CX Phone System is a software private branch exchange based on the SIP (Session Initiation Protocol) standard to allow calls via the public switched telephone network (PSTN) or via Voice over Internet Protocol (VoIP) services.
The SIP Witch is being developed since the 10th of August 2007 by David Sugar within the GNU Telephony project. The first version was 0.1.0. Version 1 was released on May 14, 2011. It is being used as a component of GNU Free Call, which is supposed to be an alternative to Skype. [1]
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.