Expo 2020 - Dubai, UAE
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 Sangoma NetBorder SS7 VoIP gateway appliance is a cost-effective, turnkey solution offering from 4 to 256 T1/E1 lines at a single point code, and powerful transcoding capabilities. Up to 8 T1/E1 lines are available in a 1U box, up to 32 lines in a 2U box.

Sangoma NetBorder SS7 VoIP Gateway product is ideal for applications such as connecting a private branch exchange to the legacy telephone network or providing multiple points of presence to a VoIP network.

The Sangoma SS7 Media Gateway provides full call control routing for SS7 traffic without the need for third party media gateway controllers or protocol converters. Full inter-working is supported across all VoIP and TDM protocols simultaneously, allowing this single multi-protocol TDM to VOIP gateway to be deployed interconnecting differing networks.

The compact, all-in-one design reduces footprint and eliminates the need to source multiple network components to handle media, signalling and routing.

SIGTRAN and MEGACO allows a distributed solution across multiple points of presence where SS7 Interconnect is required.

SNMP & Radius allows monitoring and management of NSG via both of these industry standards. A GUI provides convenient access to most configuration, monitoring and management functions, while a command line interface provides full access to management functions with a minimum of bandwidth consumption.


PSTN Protocols:

  • SS7-ISUP: ITU, ANSI, Bellcore, UK, China, India, SPIROU (France), Russian variants
  • Up to 16 A or F signalling links
  • Up to 16 Originating Point Codes
  • Up to 16 Destination Point Codes
  • Up to 16 Linksets
  • ISUP relay for larger configurations

PSTN Interfaces:

  • Up to 64 E1/T1 (960 ports) per server, available in these configurations:
  • 4 E1/T1 in 1U appliance
  • 8 E1/T1 in 1U appliance
  • 16 E1/T1 in 2U appliance
  • 32 E1/T1 in 2U appliance
  • Extend capacity up to 256 T1/E1 with relay feature
  • STM-1/OC-3 available
  • RJ-48 Connectors

VoIP Protocols:

  • SIP V2/RFC3261
  • SCTP RFC 2960
  • Megaco/H248
  • H.323

Codec Transcoding:

  • Any-to-any
  • No combination or loading restrictions
  • AMR
  • G.711
  • G.711.1
  • G.722
  • G.722.1
  • G.722.2 (AMR-WR)
  • G.723.1
  • G.726
  • G.729A
  • G.729AB
  • GSM-FR
  • iLBC
  • L8 (Linear 8K)
  • L16 (Linear 16K)
  • T.38 (fax)

Echo Cancellation:

  • G.168-2002 with 128ms tail
  • Jitter buffer

DTMF Detection and Generation:

  • RFC2833 Tone relay
  • In-band
  • DTMF detection and generation

Call Routing:

  • Flexible XML-based dial plan and routing rules
  • Any-to-any routing

Management and Configuration:

  • Web GUI
  • Command line interface
  • Call detail records in XML format
  • Detailed logs with user configurable file size and auto-rotation
  • SNMP
  • Radius
  • System backup/restore/copy


  • Per-call tracing (history and/or live)
  • Signalling capture tools
  • Command line interface
  • GUI

Session Management and Billing:

  • SIP peer availability polling
  • RTP inactivity monitoring, RTCP
  • CDR generation (RADIUS and text file)

Network Interfaces:

  • 2 RJ-45 Ethernet ports

1 for VoIP

1 for management interface

1U Appliance

4 USB ports in the back

  • 2U Appliance

4 USB ports

2 in the front

2 in the back


  • 1 DVI output port
  • AC Power:
  • 250W universal for 1U solution
  • 350W universal for 2U solution
  • DC 400W -48V for 2U (Special Order)


  • 1U : 480.4(W) x 474(D) x 44(H) mm; 19”(W) x 18.7”(D) x 1.7”(H)
  • 2U: 482(W) x 441.6(D) x 88.4(H) mm; 19”(W) x 17.4”(D) x 3.5”(H)


  • All-in-one turnkey solution
  • Up to 16 SS7 Signaling links SIP RFC3261
  • Supports a wide range of VoIP and wireless codecs
  • Sophisticated call routing via XML scripts
  • Distributed architecture and signaling relay
  • 1U

Contact us:

Address: Office 109B, Dubai Tower
Baniyas Sq., Deira, Dubai, UAE.
Telephone:  +971 4 2277586
Mobile:  +971 50 7439079
(whatsapp) +971 55 4886092
All rights reserved for SENA Co.
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