| Introduction
VoiceMaster offers a full scale H.323/SIP
compliant gatekeeper. This is the only gatekeeper in the industry
that allows dynamic call control through a proprietary calling time
authorization mechanism. Some of the key functionalities of the
Gatekeeper include:
Authentication
- IP Address of the calling station
- Username (H323ID) of the calling station
- PIN Number/Tech-Prefix of the callings station
- Calling ID (ANI) of the calling station
- Any combination of the above
The authentication is flexible enough to allow
dynamic call authentication in environments where the clients change
dynamically
their IP addresses (such as IP Phone and Microsoft NetMeeting (TM)
clients). The easiest way to implement dynamic client authentication
is PIN Number/Tech-Prefix
based. Simply use the standard PIN numbers to prepend to the phone
number.
Authorization
- Optimized Routing includes among other things LCR (least cost
routing algorithm)
- Preferred Provider Routing to allow managers to explicitly select
their termination endpoints and providers
- ASR (Average Success Rate) routing to allow dynamic selection
of providers with better ASR rates
- Route Fail over
- Authorization is dynamic and change as you change rates dynamically
to provide a single point of entry to your routing system. By
inserting new rates you also change the billing and the routing
of each call.
Dynamic Call Management
- Dynamic cal timing for single calls
- Support for wholesale multiple call timing
- Support for concurrent calls
- Dynamic call disconnect
VoiceMaster offers the only gatekeeper in the industry
that will disconnect a call after its predefined time has elapsed.
This unique dynamic call management functionality quarantines that
all wholesale and user accounts are billed dynamically against calls
in progress.
Gatekeeper Modes of Operation
- Direct/static Mode to allow call resolution
without RAS message control. This mode will allow number translation
and dynamic call control given that the participating gateways
support canMapAlias attribute.
- Routed Mode to allow direct control of RAS
messages with very low level of bandwidth utilization. This mode
allows number translation and dynamic call control for gateways
that do not support canMapAlias attribute.
- Proxy Mode to allow full RAS and RTP data transfer
for gateways behind NAT or gateways that want to keep their identity.
This bandwidth intensive mode fully controls the RAS and Q.932
data streams and supports number translation and dynamic call
control.
Optimized Routing Call Management
Optimized Routing call management is used when the
company wants to optimize the call termination costs in real time.
The algorithm allows dynamic call handling and call routing to select
the most cost efficient termination point or provider. It requires
the dialing plan management and gatekeeper functionality to be highly
integrated to allow all dialing plan changes to be propagated to
the underlying gatekeeper in real-time.
Compatibility with other vendors
- Cisco gateways and gatekeepers
- Lucent gateways
- Quintum gateways and gatekeepers
- Clarent gateways
- H323/SIP complaint gateways and gatekeepers
- IP Phones
- Lucent PIN capability to authorize Lucent PIN users
|
