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RealEast® Softswitch RSF1000
 




Advantages of RealEast® Softswitch RSF1000

RealEast® Softswitch RSF1000 (RSF1000) is the best-of-breed Softswitch class 4/5 carrier grade solution, designed to improve the performance of provider's VoIP network. The system provides such possibilities as:

RealEast╝ Softswitch RSF1000
  • Simultaneous and transparent support of SIP and H323 protocols
  • Optional RTP-proxying
  • VoIP equipment support, with NAT
  • Authentication of VoIP equipment
  • Flexible routing
  • Billing information over standard interfaces
  • System management over web-interface
  • SNMP support for statistics and trap sending
  • Distributed architecture
  • Automatic software re-start facilities when failure

Competitive benefits for service providers:

  • Market leading ROI
  • Simplify the management processes and network configuration changes of VoIP equipment
  • Unify equipment supporting different protocols (or dialects of one protocol)
  • Manage concentration and routing processes of VoIP traffic
  • Centralize authorization and billing tasks of VoIP calls in one point
  • Hide the network structure from third parties, if necessary
  • Possibility to implement value-added services such as: calling card system (RealEast® CCP) and IP-Centex system (RealEast® IP-Office) using additional software packages from RealEast Networks

Typical Applications

Being a flexible system, RSF1000 may be used in variety of telecom applications such as:


The "Brain" of your VoIP Network

Provider gets possibility to easily manage its VoIP gateways, as well as to access useful features, including:

  • Management of authorization rules of VoIP-gateways
  • Setting up call routing rules
  • Provisioning of compatibility for H323 and SIP- equipment of various vendors
  • Security and load planning of VoIP-traffic by using optional RTP-proxying
  • Communication with your billing system using RADIUS protocol
RealEast® Softswitch RSF1000

Figure 1: RealEast® Softswitch RSF1000 class4 diagram

Network Security

When using RTP-proxying RSF1000 provides a single entry point for VoIP traffic to its clients and other providers. This possibility provides reliable methods of disguising network structure thus protecting commercial interests of the parties. Unified IP address for both parties available (clients - provider). All other devices are hidden.


Integration of Equipment With Support of Different Protocols

One of the most important features of RSF1000 is its ability to support widely accepted signaling IP-protocols - SIP and H323. The system provides transparent converging of one protocol into another, thus allowing performing calls from one type of equipment to another.

Note: The system has integrated support of SIP to H323 conversion for H323 "slow start" variant as well.


Specifications


Supported protocols
1 SIP ( RFC 3261)
2 H323 v4 (H.245 v 7, H.225 v 4) with/without fast start
3 RTP/RTCP (optional proxying)
4 T .38, T .120
5 Protocol conversion (SIP and H.323)
Authentication
1 Authentication of SIP and H.323 equipment by IP address
2 Authentication of SIP and H.323 equipment by Caller ID prefix and IP address
3 Authentication of SIP and H.323 equipment by Callee ID equipment and IP address
4 Authentication of SIP and H.323 equipment by prefix combination of the Caller and Callee numbers and IP address
Call routing
1 By telephone number prefix of the Callee-ID
2 By telephone number prefix of the Caller-ID
3 By prefix combination of the Callee and telephone number prefix of the Caller
4 Least Cost Routing (LCR)
5 By Quality of service (QoS)
6 By ASR
7 By ACD
8 Reroute By Busy, Timeout, Service Unavailable (34) and if Call Rejected
9 Possibility to set up flag to stop termination search (Hunt Stop)
Forming telephone number to dial
1 Possibility to remove given number of digits from the Caller ID
2 Add given number of digits to the Caller-ID
3 Possibility to remove given number of digits from the Callee ID
4 Add given number of digits to the Callee-ID
Billing information output
1 RADIUS protocol (CISCO Attribute 44)
2 CDR record in text file
Configuration and administration
1 VoIP gateway control and routing by Web-interface
2 View statistics and call control on-line by embedded Web-interface
3 Output of information by SNMP-traps mechanism
4 PC parameter control by text configuration files
Operation system
1 Linux
2 Windows 2000

For additional information please contact us at products@realeastnetworks.com


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