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BorderNet™ 500 Gateways
 

Dialogic® BorderNet™ 500 Gateways are turnkey appliances that can enable the rapid deployment of new SIP-based communications services to enterprise customers by providing a flexible means to deliver SIP services from public IP networks to private enterprise IP networks and their resident communications systems.

BorderNet 500 Gateways supply any-to-any connectivity and call routing for connection to SIP trunks or PSTN trunks and virtually any on-premise PBX, including IP-PBXs, hybrid PBXs, and legacy TDM PBXs, along with integrated enterprise session border control (SBC) features. SBC features include Network Address Translation (NAT) traversal, network-edge security, and a wide variety of SIP controls for interoperability. By defining a distinct and secure demarcation point or border for SIP services between public and private networks, the SIP service can become both more manageable and reliable.

 

  Features   Benefits
  Any-to-any connectivity and call routing   Provides flexibility in connecting to a wide variety of services and equipment, including SIP trunks, PSTN trunks, and legacy, hybrid, and IP PBXs
       
  Extensive interoperability testing with SIP service providers and PBX manufacturers   Delivers a high degree of confidence that BorderNet 500 Gateways will work effectively with a wide variety of vendor interfaces and equipment
       
 

Robust SIP security features

  Creates a secure demarcation point for an enterprise at the network edge to fend off malicious outside threats
       
  Built-In SIP Proxy to enable firewall and NAT traversal   Allows an enterprise to connect to a SIP trunk or SIP service
       
  Detailed call quality statistics   Enhances the ability to troubleshoot voice quality issues
       
  Optional software modules   Allows an enterprise to tailor its network edge solution to user needs with added QoS, enhanced security, remote access, and primary SIP endpoint registration
       
  T.38 Fax over IP (FoIP) at V.34 speeds   Includes high speed, reliable FoIP that reduces expenses by decreasing the time needed to transmit/receive fax messages
       

 

  Technical Specifications    
  Server Type  
  • Nexcom NSA 3110
       
  Processor  
  • E1500 Celeron, 2.2 Ghz
       
 

Memory

 
  • 1GB RAM1066 DIMM DDR3
       
  Hard disk subsystem  
  • Hitachi 500GB (24X7 rated)
       
  Network interface  
  • 4x 10/100/1000 Base-T Ethernet ports
  Protocol support  
  • ISDN BRI: DSS1 (Euro-ISDN), NI-1, 5ESS, 1TR6, INS Net 64, VN3, CT1, QSIG
  • E1 ISDN: ETSI-DSS1 (EuroISDN), INS-1500 (Japan), QSIG
  • E1 CAS: MFR2
  • T1 ISDN: NI-1, 4ESS, 5ESS, DMS100,
  • QSIG T1 CAS: RBS
  VoIP services  
  • SIP methods: ACK, BYE, INVITE, NOTIFY, REFER, CANCEL, OPTIONS, REGISTER
  • Configurable IP transport layer UDP or TCP
  • Number normalization and manipulation of Called/Calling/Redirected Number
  • Call Routing based on Called/Calling/Redirected Number, PSTN Interface, and/or SIP Peer
  • Call Hold/Retrieve (for example, Re-Invite mapping towards ISDN) PSTN-side Call Transfer (REFER points to PSTN)
  • Call Diversion
  • Message Waiting Activation/Deactivation
  • Call Redirection via 302 Moved Temporarily
  • Simplified Number Normalization based on PSTN connection parameters
  • Number Manipulation using Regular Expressions
  • Configurable Cause Code Mapping
  • Clear Channel Fax
  • Clear Channel Modem
  FoIP (T.38) services  
  • T.30 Fax Group 3 up to 33.6 kbps using T.38 real-time FoIP
  • Fax compression MH, MR, MMR
  • Error Correction Mode (ECM)
  Additional SIP features  
  • SIP Proxy and Registrar
  • SIP Connect Compliant Security
  • TLS and SSL authentication
  • SRTP (Secure Real-time Transport Protocol)
  • SIPS (Secure SIP)
  • Supported ciphers: DH, ADH, AES (128-256 bits), 3DES (64 bits), DES (64 bits), RC4 (64 bytes), RC4 (256 bytes), MD5, SHA1
  Reliability  
  • Load balancing and failover on PSTN side
  • Load balancing and failover on SIP side (optionally uses OPTIONS for keep-alive check)
  • Alive check for active calls on SIP side via SIP session timer (RFC4028)
  Call routing  
  • TDM-to-TDM
  • TDM-to-SIP
  • SIP-to-TDM
  • SIP-to-SIP
  Media processing features  
  • DTMF generation and recognition (in-band)
  • DTMF relay, RFC2833
  • Echo Cancellation as per G.168 standard with up to 256 ms echo tail (depending on media gateway interface)
  • Voice Activity Detection and Comfort Noise Generation
  IP Media CODEC features  
  • IP Real-time Transport Protocol (RTP)
  • RTP profile name RTP/AVP
  • RTP event (RFC2833) for DTMF, fax, and modem tones
  • G.711 CODEC, 64 kbps (64 kbps, A-law, µ-law)
  • G.726 (16, 24, 32, and 40 kbps)
  • G.729 CODEC (requires additional license from Dialogic)
  • GSM full rate CODEC
  • iLBC CODEC
  • Comfort Noise (RFC3389)
  • Configurable packetization time between 20 ms and 200 ms (iLBC only between 20 ms and 30 ms)
  Management  
  • Configuration via web GUI (HTTP or HTTPS) or CLI
  • SNMP for monitoring
  • Logging to PCAP file, SYSLOG
  • Radius interface
  Physical dimensions  
  • Height: 44 mm (1U)
  • Width: 426 mm
  • Depth: 365 mm
  Power supply  
  • 200 W ATX Supply
       

 


 

* All document will in Portable Document Format (PDF).

For details : sales@icg-corp.com

 

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