sipX Architecture

The sipX project is about developing the most feature rich and standards compliant SIP communications infrastructure for Enterprise use in a community organized open source effort. The sipX architecture is modular and consists of three main building blocks:

• sipX Communications Server
• sipX Media Server
• sipX Configuration Server

While sipX packages these components to function as a SIP PBX, each server can also be used standalone.

1. The sipX Communications Server

The sipX Communications Server provides the core PBX call control and routing functions. It provides the signaling infrastructure for sipX, including SIP proxy, registry, location, redirect, authentication, and status functions using four modules:

• Proxy Server - Directs all messages to various system elements using forwarding rules triggered by the content of the SIP message. Calls may require the location services of the Contact Server module or the authentication and authorization services of the Security Server module.
• Contact Server - Provides a SIP location service that determines the current methods to contact a particular phone number, ensuring that a call can be routed to the correct phone, gateway, or the Media Server. To perform this function, the Communications Server maintains a dynamic database of active SIP registrations that map phone numbers to the IP addresses of IP phones. The Contact Server module also contains a set of static XML-based mapping rule files that describe routing rules, dial plans, gateways, hunt group definitions and aliases for E.164 numbers, IP addresses, and URL addresses. Customized manipulation of the XML mapping rules permits the server to be extended in powerful ways. This combination of dynamic and static information allows Contact Server to determine the current “contact address” for a SIP Invite message.
• Security Server - Determines whether a SIP message needs to be authenticated via SIP Challenge/Response and whether the requested action is within the permissions scope of the requesting user. SIP requests that are intended for a protected resource, such as a SIP/PSTN gateway, must provide a set of valid authentication credentials and call admission permissions before the call is allowed. Security Server uses a data set of authentication rules to validate the credentials for the user. After authenticating the user, Security Server consults a permissions data set to determine whether the user can place such calls. Once Security Server validates the user’s authorization, the call is placed.
• Status Server - Provides a generalized SIP Subscribe/Notify mechanism that is widely used throughout the system for various purposes, such as message waiting indication, configuration change notifications, and other functions.

sipX Call Routing

sipX Communications Server supports all SIP dialing plan standards, including SIP URL and email call addressing. Once a call is initiated by a SIP endpoint, a SIP Invite message is sent over the IP network to the Communications Server, where the four modules perform their respective roles for call setup before forwarding the message to the destination phone, gateway, or Media Server. Once the two endpoints complete their call Communications Server re-enters the session to assist in call teardown.
The ability of the Communications Server to apply selective routing rules to particular dialed numbers means that sipX is not only a PBX but it can also mimic traditional CLASS 4 toll bypass functions. Toll-bypass calling is a key benefit of the sipX system

sipX System Management and Administration

Management of Communications Server operations is provided by the Configuration Server component of sipX, which centralizes management and eliminates the need to perform administrative tasks on multiple systems or a dedicated management terminal.

sipX Scalability and High-Availability Configurations

Communications Server employs the same Domain Name System (DNS) round-robin techniques used in large-scale Web server farms to deploy failover and redundant components, such as additional servers, to create a system that is highly scalable and inherently reliable. The DNS technique for redundant components applies to both single sites and multi-locations, avoids the limits of Windows clustering, and eliminates single points of failure.
When a SIP phone, soft phone, or gateway attempts to make a call, it performs a DNS lookup to determine the preferred protocol (TLS over TCP or UDP) of the available servers within the domain specified in the SIP URL (or the configured next-hop SIP server), and then selects one of the servers from the supplied list of servers.
The ability of phones to sequentially seek servers until a working server is successfully contacted forms the high availability foundation of sipX. System components can “fail over” to any number of SIP servers to prevent call blockage. This addresses the risk of server inaccessibility due to load, server failure, or a network segment outage. Alternately, the same technique can be used to distribute the SIP messaging load hierarchically to a number of geographically scattered sipX servers.

2. The sipX Media Server

sipX Media Server is an interactive voice response IVR solution providing capabilities for auto attendant, automatic call routing, and voicemail as well as an open architecture for customization and advanced functionality.
As one of the three primary components of the sipX IP PBX, Media Server works closely with sipX Communications Server and Configuration Server to provide an integrated PBX solution.
sipX Media server is a native SIP application built using the Open VXI VXML interpreter, which executes VXML logic to play and record audio prompts, messages, and scripts that are stored as files on the server. Media Server permits customization of auto attendant and IVR features by simply modifying the VXML scripts with any third-party commercial VXML authoring application or by recording new audio files.
Native SIP support permits a basic Media Server to scale the same way sipX Communications Server scales. In addition, the application’s modular Internet architecture allows a user to distribute servers and storage on the network, with the possibility to provide high reliability, backup, and load balancing.
The Media Server component provides a voicemail system that allows end-users to retrieve their voicemail from a desk phone, an outside phone, an email program, and a Web browser on any PC with audio capability. In addition, end users can employ the Web browser interface to manage, organize, label, and store their voicemail messages.
Media Server is administered using sipX Configuration Server.

3. The sipX Configuration Server

sipX Configuration Server provides administration and management of the sipX solution as well as remote configuration for SIP phones and gateways. The Configuration Server simplifies overall system management with an intuitive browser interface and back-end systems for centralized configuration and operation of system components, phones and gateways. Built on Web server, Web browser, and RDBMS foundations, the Configuration Server software offers centralized control of the Communications Server, Media Server, user configurations, software delivery, individual phones, administrator-specified phone groups, and the routing rules that are executed by Communications Server.
Configuration Server uses a “profile generator” to turn its database of configuration information into the appropriate syntax for phones on the system. With this capability Configuration Server is able to remotely manage SIP compliant phones and gateways from potentially any manufacturer.
sipX Configuration Server includes the following four major components:

• Administrator Interface -Driven by a J2EE-based back-end web server application, this browser-based administrator interface provides facilities for adding users, devices and applications to the system, changing call handling preferences and other configuration-oriented items. It also serves as the operational console for starting and stopping sipX server components.
• End-User Interface - This browser-based interface permits end-users to control their phone settings for call forwarding, voicemail, and speed dial within parameters specified by the system administrator.
• SQL Configuration Database - Information about device and user configuration settings, dial plans, phone and gateway MAC addresses, user permissions, and other system parameters are stored in a SQL database, embedded in the Configuration Server.
• Configuration File Generation and Distribution - A “profile generator” function extracts information from the user interface and database tables to create configuration files for each system component (phones and gateways) and deliver those configuration files to the affected components by either push or pull, as appropriate.

To protect the system from unwanted intrusion, Configuration Server offers configurable security with policy-based permissions that determine which individuals have access to the various system components, from the central servers to individual phones and voicemail inboxes.
Based on the user ID log in, Configuration Server presents a different set of options to each person. For example, “superadmin” users are allowed to configure system-wide settings while end-users are only allowed to manage their own features, such as their password, call forwarding behavior, speed dial numbers, and voice mail options.