sipXecs Family of Projects

sipXecs - A native SIP based Unified Communications Solution

What is it we are trying to do?

Session Initiation Protocl (SIP) has become to real-time communications what HTTP is to Web surfing and SMP for email communication. We have set an ambitious goal for ourselves aiming at building the most interoperable, feature complete, scalable, robust and easy to use SIP unified communications system in open source.

The sipXecs solution is a new open source Enterprise unified communications solution natively based on the Session Initiation Protocol (SIP). Voice over IP (VoIP) should be easy, so sipXecs was built with that in mind. Architected as a pure SIP solution sipXecs comes complete with lots of features such as voicemail, auto-attendant, call center solution, group paging, intercom, personal attendant, find-me / follow-me, integration with Microsoft Active Directory and Exchange 2007, plug & play management, and many more.

We often get asked whether the SIP standard is mature and complete enough to implement all the features and capabilities required for a comprehensive solution. This is a legitimate questions since all the other existing commercial solutions either add proprietary extensions to the standard to "fill some gaps that otherwise could not be overcome in a standards compliant way" or merely use the SIP protocol as a transport infrastructure combined with a vendor proprietary core for call control. We think SIP is fully mature as a standard to rely on for everything that is needed to build a standards compliant unified communications solution. The sipXecs is vivid proof of this as we have yet to come accross something that cannot be implemented in a standards compliant way. 

Who do we think should use sipXecs?

A project without a clear target user group is like a speaker without an audience. Therefore sipXecs is a project with a clear focus on making something that can be used successfully for an intended purpose. sipXecs is not just an experiment, but a serious solution that aims at replacing existing PBX systems, small and large, with a better solution. 

sipXecs is architected as a fully distributed system with a centralized configuration and management application. As such all the components can run on a single server rendering a very cost effective and powerful system for the smallest offices. If configured to run on different hardware a powerful system can be created that serves thousands of users in different locations. 

sipXecs is a modular server based solution that runs on standard Linux and does not require any additional hardware as it interoperates with any SIP compliant gateway, phone or application. If offers redundancy for both servers, applications and trunk lines and therefore allows creating a system as resilient as required by the user's environment.

Why should you join the sipXecs project as a developer or tester?

If you are interested in a professional athmosphere and would like to interact with some of the industry's premier experts on the SIP protocol, the sipXecs project might be right for you. The project uses a lot of interesting technoologies and operates in a sophisticated environment using lots of tools. We are open to suggestions and good ideas and since components are fairly independent of each other you can come in and do it your way. A good example of this is the recent edition of our new group paging server, a component written in Java and using the JainSIP stack in stark contrast to the otherwise usual C++ enviroment based on the sipX SIP stack.

 sipXecs is a native SIP communications solution strictly following and implementing all the relevant SIP IETF standards. It is designed as a distributed architecture where all the components can either reside on a single server or be distributed across several server platforms.

sipXconfig - Plug & Play Web-based Management

The sipXecs open source IP PBX Configuration Server provides a browser-based interface for system administrators to configure and manage users, phones, applications, servers, and the system itself. The Configuration Server allows remote administration and configuration management of the total solution as it automatically generates configuration files for use by other sipXecs system components, including the Communications Server, the Media Server, and end-user phones from various vendors. Additional phones and gateways can be added easily using a plug-in framework.

The sipXecs Configuration Server provides a SOAP-based Web Services interface for easy integration into Web Portals. In addition, it allows importing of configuration data from a "cut-sheet" such as an Excel spread sheet. All configuration data is stored in a PostgreSQL database and automated mechanisms are provided for backup & restore.

A persoanl user configuration portal is offered for every user on the system that allows user self-control over key features of the IP PBX system. Such features include voicemail management, settings for unified messaging, configuration of a personal auto-attendant, personal language settings for the user's voicemail inbox, speed dial and directory configuration, presence management, as well as a personal call history.

Detailed reporting is offered for Call Detail Records (CDRs) representing detailed call statistics. The Call Center (ACD) system offers both real-time and historic reporting provided by the sipXecs configuration management system. Report data can also be exported. 

All the phones and gateways are managed plug & play including auto-discovering devices on the LAN. While all configuration is auto-generated by default by the system, the admin is given the opportunity to manipulate every single parameter in such connected devices. Therefore, it is no longer necessary to login to phones and gateways and learn their respective configuration interfaces and mechanisms.

sipXtapi - A powerful high level SIP client toolkit

sipXtapi is a very powerful API and toolkit that provides SIP end point (phone and media server) functionality. sipXtapi provides application developers with a very simple to use high level API that does not require any knowledge of the SIP protocol. sipXtapi allows developer to focus on telephony, video, presence and IM functionality without having to touch or worry about SIP signalling.

sipXtapi is particularly suited for:

• Softphones, Embedded Systems, Media Servers and Peer-to-Peer Solutions
• Media Enabled End Points
• Secure SIP Applications using TLS, S/MIME and SRTP
• Multi-platform Support: Windows, Mac OS X, Linux (VxWorks, WinCE and Symbian in progress)

sipXtackLib - A highly portable SIP stack in C++

sipXtack is a highly portable, standards compliant, highly interoperable, and feature rich SIP stack. The sipX SIP stack supports, RFCs 3261, 3263, 3264, 3265, 3420, 3428, 3515, 3581, 3842, 3856, 3863, 3891 and many others. The sipX SIP stack has been tested at all 17 of the SIPit SIP interoperablity events since the events were first held in Apr. 1999.

The sipX SIP stack is particularly suited for:

• SIP Clients, Softphones, Embedded Systems, Media Servers, Gateways
• SIP Peer-to-Peer (P2P) Applications
• Applications requiring proxying functionality
• Voice, Video, Presence and Instant Messaging
• Secure SIP Applications requiring SIPS, TLS, S/MIME or SRTP
• Multi-platform Support: Windows, Mac OS X, Linux (VxWorks, WinCE and Symbian in progress)

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