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| Developer(s) | Creacode Ltd. |
|---|---|
| Initial release | 2006 |
| Operating system | Microsoft Windows |
| Available in | English, Turkish |
| Type | VoIP |
| License | Proprietary |
| Website | Creacode Ltd. |
Creacode SIP Application Server is a SIP call controller, SIP IVR, SIP Registrar and SIP NAT traversal software product.
Creacode SIP Application Server's javascript-like, event-driven scripting language (CCS) and internal APIs allow real-time control on SIP signals for building complex voice services in VoIP networks. It communicates with external web applications, RADIUS servers or ODBC compliant relational databases in order to save or retrieve data associated with each call session during execution of call services. IVR feature supports to play and record G711U, G711A and G729A audio codecs. Creacode SIP Application Server's core is written in unmanaged C++ code to meet high performance and scalability requirements. It runs on 32/64 bit Windows operating systems.
Creacode SIP Application Server is compliant with RFC 0821, 1325, 2327, 2616, 2833, 2865, 2866, 2976, 3261, 3264, 3515, 3550, 3565, 4566.
The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating real-time sessions that include voice, video and messaging applications. SIP is used for signaling and controlling multimedia communication sessions in applications of Internet telephony for voice and video calls, in private IP telephone systems, in instant messaging over Internet Protocol (IP) networks as well as mobile phone calling over LTE (VoLTE).
Interactive voice response (IVR) is a technology that allows humans to interact with a computer-operated phone system through the use of voice and DTMF tones input via a keypad. In telecommunications, IVR allows customers to interact with a company’s host system via a telephone keypad or by speech recognition, after which services can be inquired about through the IVR dialogue. IVR systems can respond with pre-recorded or dynamically generated audio to further direct users on how to proceed. IVR systems deployed in the network are sized to handle large call volumes and also used for outbound calling as IVR systems are more intelligent than many predictive dialer systems.
HCL Sametime is a client–server application and middleware platform that provides real-time, unified communications and collaboration for enterprises. Those capabilities include presence information, enterprise instant messaging, web conferencing, community collaboration, and telephony capabilities and integration. Currently it is developed and sold by HCL Software, a division of Indian company HCL Technologies, until 2019 by the Lotus Software division of IBM.
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A VoIP phone or IP phone uses voice over IP technologies for placing and transmitting telephone calls over an IP network, such as the Internet, instead of the traditional public switched telephone network (PSTN).
Gizmo5 was a Voice over IP communications network and a proprietary freeware soft phone for that network. On November 12, 2009, Google announced that it had acquired Gizmo5. On March 4, 2011, Google announced that the service would be discontinued as of April 3, 2011.
QuteCom was a free-software SIP-compliant VoIP client developed by the QuteCom community under the GNU General Public License (GPL). It allows users to speak to other users of SIP-compliant VoIP software at no cost. It also allows users to call landlines and cell phones, send SMS and make video calls. None of these functions are tied to a particular provider, allowing users to choose among any SIP provider.
Ekiga is a VoIP and video conferencing application for GNOME and Microsoft Windows. It is distributed as free software under the terms of the GNU General Public License. It was the default VoIP client in Ubuntu until October 2009, when it was replaced by Empathy. Ekiga supports both the SIP and H.323 protocols and is fully interoperable with any other SIP compliant application and with Microsoft NetMeeting. It supports many high-quality audio and video codecs.
FreeSWITCH is a free and open-source application server for real-time communication, WebRTC, telecommunications, video and Voice over Internet Protocol (VoIP). Multiplatform, it runs on Linux, Windows, macOS and FreeBSD. It is used to build PBX systems, IVR services, videoconferencing with chat and screen sharing, wholesale least-cost routing, Session Border Controller (SBC) and embedded communication appliances. It has full support for encryption, ZRTP, DTLS, SIPS. It can act as a gateway between PSTN, SIP, WebRTC, and many other communication protocols. Its core library, libfreeswitch, can be embedded into other projects. It is licensed under the Mozilla Public License (MPL), a free software license.
Telesoft Technologies is a privately held UK based technology company which develops cyber security, telecoms mobile products and services and government infrastructure. Telesoft has operations in USA, UK and India.
Jami is a SIP-compatible distributed peer-to-peer softphone and SIP-based instant messenger for Linux, Microsoft Windows, OS X, iOS, and Android. Jami was developed and maintained by the Canadian company Savoir-faire Linux, and with the help of a global community of users and contributors, Jami positions itself as a potential free Skype replacement.
Voxeo Corporation was a technology company that specialized in providing development platforms for unified customer experience (self-service) and unified communications applications. Voxeo was headquartered in Orlando, Florida with main offices in Cologne, Germany; Beijing, China; London, UK and San Francisco, US.
Brekeke PBX is a SIP-based IP-PBX system. This software was previously known as OnDO PBX from 2004 to 2006. Brekeke Software, Inc. released Version 2 of this software in 2007 and changed its name to Brekeke PBX. This software allows the user to establish relations among multiple phones, to share lines among multiple phones and to provide call features such as voicemail, caller ID, call forwarding, call recording, etc. Brekeke Software, Inc. offers two versions of its Brekeke PBX software, single-tenant and Multi-Tenant version.
VaxTele SIP Server SDK is a complete development toolkit, which allows software vendors and Internet telephony service providers (ITSP) to develop SIP Server and (SIP) Session Initiation Protocol based VoIP systems for Microsoft Windows to install computer to computer voice chat, chat rooms, IVR systems, call center services, calling card services, dial/receive computer to PSTN and mobile phone calling services.
Voice Elements is a Microsoft .NET development environment for building automated telephone systems. Voice Elements was released by Inventive Labs Corporation in 2008, based on their original CTI32 toolkit. Software developers who use C#, VB.NET or Delphi use Voice Elements to write telephony-based applications, such as Interactive Voice Response systems, voice dialers, auto attendants, call centers and more.
Dial-Office IP-PBX is a SIP-based IP-PBX business phone system, first released in 2003 by Canadian telecommunications software provider Dialexia. The software allows users to connect multiple phones, share lines among several phones and implement business PBX telephone phone features such as voicemail, caller ID, call forwarding & call recording into their system. Dial-Office is also suitable for multi-office connections, connecting branches which are geographically distant from each other. Dialexia Communications, Inc. released the latest version of Dial-Office IP-PBX (4.1) in 2013. On June 3, 2014, the company's development team announced in a client newsletter that support for Dial-Office versions 3.9 and earlier would cease effective September 1, 2014.
HERO Hosted PBX is a SIP-based hosted IP-PBX business phone system, first released in 2008 by Canadian telecommunications software provider Dialexia. The HERO software allows users to connect multiple phones, share lines among several phones and implement business PBX telephone phone features such as voicemail, caller ID, call forwarding & call recording into their virtual PBX. The software is also suitable for multi-office connections, connecting branches which are geographically distant from each other. Dialexia Communications, Inc. released the latest version of HERO Hosted PBX (4.3) in 2013.
WebRTC Gateway connects between WebRTC and an established VoIP technology such as SIP. WebRTC is an API definition drafted by the World Wide Web Consortium (W3C) that supports browser-to-browser applications for voice calling, video chat, and messaging without the need of either internal or external plugins.
