Dialogic® BorderNet™ 500 Enterprise Session Border Controller

BorderNet 500 Gateway

Turnkey appliances that enable any-to-any routing for the connection of SIP Trunks and hosted SIP services.

The Dialogic® BorderNet™ 500 Enterprise Session Border Controller is a turnkey appliance 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.

The BorderNet 500 ESBC supplies 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.

Note: The BorderNet 500 ESBC was formerly known as the Dialogic® BorderNet™ 500 Gateway.

FeaturesBenefits

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 the BorderNet 500 ESBC 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
          Approvals, Compliance, and Warranty 
           Hazardous substances RoHS compliance information at http://www.dialogic.com/rohs
           Country-specific approvals 
          		Global product approvals database at http://www.dialogic.com/declarations
          Warranty Warranty information at http://www.dialogic.com/warranties

          Manuals

          Brochure

          Data Sheet

          • Dialogic® BorderNet™ 500 Enterprise Section Border Controller
            BorderNet 500 ESBC is a turnkey appliance 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.

          Technology Brief

          • Dialogic and Next Generation Public Safety

            Dialogic is a global leader in enabling the secure and seamless transition of mission-critical Public Safety networks from legacy to next-generation platforms.  For solutions from incident notification to mass reporting and coordinated response, Public Safety leaders are building on Dialogic. 

          • Security and Interoperability Issues to Consider When Enabling SIP Services for the Enterprise
            If you are thinking about deploying a SIP trunking service, there are several issues that you may want to consider: secure firewall traversal, SIP interoperability and security, VoIP service demarcation, legacy PBX integration, and Fax over IP (FoIP) support. This technology brief discusses these topics and introduces the Dialogic® BorderNetTM 500 Gateways as network edge products that can help you deliver SIP trunking services smoothly and efficiently in your enterprise environment.
          • New Video Conferencing Demo from Dialogic
            The Dialogic® Video Conferencing Demo provides several examples of how video conferencing can be deployed. You can download this demo free-of-charge and use it for 45 days to demonstrate video conferencing solutions that support the latest endpoints, including smartphones and tablets.

          White Paper

          • SIP Trunking: Deployment Considerations at the Network Edge
            This paper provides an overview of SIP trunking and the benefits it can bring to the enterprise. It also provides information about how to address issues at the enterprise network edge when deploying a SIP trunking service.
          • SIP Trunking: Enabling Wideband Audio for the Enterprise
            This white paper focuses on how an enterprise with SIP trunking can use wideband audio for internal communications, communications with other enterprises, and communications with mobile users to create a better user experience.
          • LTE Drives Opportunities for Cloud-Based Mobile Video Services, Operators, and Providers
            As LTE networks are deployed, broadband access from smartphones will become nearly ubiquitous, allowing mobile users more access to new services in areas such as entertainment, advertising, and video-enabled call centers. Businesses can benefit by providing these new services as "cloud-based" to enable fast, cost-effective, accessible, and scalable deployment for mobile users.

          Use the Purchase page linked below to find partners to purchase Dialogic products. The order codes below are for your reference when making a purchase.

          Purchase

          ProductOrder CodeDescription
          BN500IP306-422ESBC only, 0 TDM Ports, 25 SIP-to-SIP Sessions
          BN508LS306-423ESBC + Gateway, 8 TDM Ports, 8 SIP-to-SIP Sessions
          BN508BRI306-424ESBC + Gateway, 4 TDM Ports, 8 SIP-to-SIP Sessions
          BN501PRI306-425ESBC + Gateway, 30 TDM Ports, 30 SIP-to-SIP Sessions
          BN504PRI306-426ESBC + Gateway, 120 TDM Ports, 120 SIP-to-SIP Sessions
          BN504PRIV34306-481ESBC + Gateway, 120 TDM Ports, 120 SIP-to-SIP Sessions
          BN500SWRCMM01-201-01Remote SIP Connectivity Module, N/A TDM Ports, N/A SIP-to-SIP Sessions
          BN500SWQOSMM01-202-01QoS Module, N/A TDM Ports, N/A SIP-to-SIP Sessions
          BN500SWESMM01-203-01Enhanced Security Module, N/A TDM Ports, N/A SIP-to-SIP Sessions
          BN500SWVSMM01-204-01VoIP Survivability Module, N/A TDM Ports, N/A SIP-to-SIP Sessions
          BN500SWSRMM01-205-99SIP Registrar Module, N/A TDM Ports, N/A SIP-to-SIP Sessions
          BN500SWASTCM01-206-01Additional SIP Traversal Channel, 0 TDM Ports, 1 SIP-to-SIP Session