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Telephony SIP Proxy Server

 
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INTRODUCTION

The JIP6500V2 SIP Telephony Server is the best choice to your convergence VOIP network which covert the requirements from enterprise to service provider. With built-in rich telephony services, JIP6500V2 enables traditional PABX features to your VOIP convergence platform. Also you can easily upgrade the license or provide the high available service according to the growth of your business without any hardware changes.

 

KEY FEATURE

Intelligent Call Routing
  JIP6500V2 provides multiple service routing polices to meet different service providers' requirements (e.g. load balancing,priority, most idle etc.) It enables service provider to tell how to route the call depending on the call results or predefined rules. The incoming prefix match and outgoing prefix insert provides a very easy way to manage your VOIP exchange service.
Easy to Configure and Management
  Full web management interface make you to manage your JIP6500V2 anywhere of the world. You don't need remember the command lines or operate it on the specified console. Also the system event notice
features keep you the system status updated remotely.
NAT On-Demanded Traversal
  Due to the lack of IPV4 address, a lot of customer is using NAT for their network. JIP6500V2 provides the NAT on-demanded traversal which will only route the voice when needed. It saves the bandwidth and provides better voice quality compare to route each call voice back to server. No CPE modification is required.
Voice NAT/Firewall Router
  With built-in SIP and voice routing features, JIP6500V2 provides a secure and easy way to migrate your Voice IP PBX solution. It acts as a NAT router and firewall role which voice RTP port is only opened when SIP signaling is established successfully.
Rich Telephony Service
  The JIP6500V2 provides build-in rich set of telephony service which enables the service provider quick time to market to delivery their service to their customers. By cooperating to Justec IPCentrex 6850, the service provider can provide Announcement, Auto Attendant, VMS, CRBT etc. immediately.
Multiple Access Protocol
  With provided SIP TCP and UDP protocol, JIP6500V2 can acceptboth type of signals and do the conversionwhen needed. For each protocol, JIP6500V2 cansupport up-to 3 service ports which enabling to receiveSIP service anywhere of world.
High Availability Redundant
  JIP6500V2 provides high availability VOIP service by using active and stand-by redundant technologic which provides hot standby and hitless fail-over for stable call to reach mission-critical service requirement. It keeps your service continues running.
Microsoft Live Communicator Server 2005 Integration
  JIP6500V2 can work with Microsoft Live Communication Server 2005 as a total solution to meet the enterprise communication requirement. Without any extra settings in LCS 2005,JIP6500V2 is able to provide all features that JIP6500V2 provided. Both of your SIP phone and Office Communicator can ring to talk together.
   

ORDERING INFORMATION

JIP6500V2 Telephony SIP Proxy Server

 

 
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