SysMaster Gatekeeper

Details

Dynamic Call Management

SysMaster VoIP Gatekeeper is the only product in the industry that will disconnect a call after its predefined time limit has elapsed. This unique functionality guarantees that all wholesale and user accounts are billed dynamically against calls in progress. Other features of the gatekeeper include dynamic call timing for single calls and wholesale multiple call timing.

Multiple Routing Mechanisms

SysMaster VoIP Gatekeeper supports multiple routing mechanisms such as Optimized Routing (including least cost routing), Preferred Provider Routing, Average Success Rate (ASR) Routing and Route Fail Over. That feature enables providers to select the most profitable and high quality routes for each call, increasing both ASR and profit margins.

Unlimited Dialing Plans/Routing Tables

SysMaster VoIP Gatekeeper also supports unlimited number of inbound and outbound dialing plans/routing tables as well as full scripting language for number translation. Dialing plans can be bound to PSTN lines, network IP addresses/network segments, or end points and can be time-based.

Multiple Authentication Methods

SysMaster VoIP Gatekeeper supports multiple methods for subscriber authentication. Among the supported methods are PIN, ANI, DNIS, ANI/PIN, Tech-prefix, User Name (H.323) and IP address.

Dynamic Authentication

SysMaster VoIP Gatekeeper also supports dynamic authentication to provide a single point of entry to the system in environments where clients change dynamically their IP addresses (such as IP Phones and Microsoft NetMeeting ™ clients). The functionality is implemented through pre-pending client PIN/Tech-prefix to the dialed phone number.

Proxy Mode Support

SysMaster VoIP Gatekeeper can operate in proxy mode, allowing RAS and RTP data packet transfer for gateways behind NAT or for gateways that want to keep their identity. In this bandwidth intensive mode the gatekeeper fully controls the RAS and Q.932 data streams and supports number translation and dynamic call control.

Routed Mode Support

SysMaster VoIP Gatekeeper is also capable of operating in routed mode with low level of bandwidth utilization. In such mode the gatekeeper performs direct control of RAS messages and allows number translation and dynamic call control for gateways that do not support canMapAlias attribute.

Static Mode Support

In static mode configuration SysMaster VoIP Gatekeeper resolves calls without RAS message control. In such mode the gatekeeper allows number translation and dynamic call control if the participating gateways support canMapAlias attribute.

Scalability

SysMaster VoIP Gatekeeper is designed to accommodate the growth of service providers' networks. In H.323 setups, the product can scale to handle up to 20,000 calls in static mode, 10,000 calls in routed mode, and 7000 in proxy mode. In SIP setups, the gatekeeper can handle up to 10,000 calls in routed mode and up to 7000 calls in proxy mode.

Full Compatibility with Other Vendors

SysMaster VoIP Gatekeeper supports any standards compliant VoIP equipment. Among others, the gatekeeper supports Cisco and Quintum gateways and gatekeepers, Lucent and Clarent gateways as well as any H.323/SIP compliant end points.