Effortless connection between PSTN SS7/TDM and VoIP networks
- Connect calls between PSTN and VoIP networks
- Integrated media gateway, signalling gateway and media gateway controller
- Compatible with distributed architecture
- Scalable, carrier-grade
- Highly interoperable with other system types
- Supports a wide range of codecs for VoIP, PSTN and wireless
- Supports a wide range of SS7 and VoIP signalling protocols
- Sophisticated call routing
The 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.
This 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.
Applications
- Alternative carrier solutions
» Least-cost routing
» SIP–SS7 interworking - ISUP-based caller ring back tone
- Pre-paid and overseas dialing
- Multiple points of presence
- 2U SS7 gateway
- 1U SS7 gateway
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 32 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 over 960 ports and single server via ISUP relay feature
- RJ-48 Connectors
VoIP Protocols:
- SIP V2/RFC3261
- SIGTRAN M2UA RFC 3331
- SCTP RFC 2960
- Megaco/H248
- H.323
| RFC 2246 | Transport Layer Security (TLS) for SIP |
| RFC 2327 | Session Description Protocol (SDP) |
| RFC 2976 | SIP Info for digit transmission (#,*) and interworking DTMF |
| RFC 3261 | SIP Basic |
| RFC 3263 | Locating SIP servers for DNS lookup SRV and A records |
| RFC 3264 | SDP Offer/Answer Model |
| RFC 3265 | SIP Subscribe/Notify |
| RFC 3326 | SIP Reason Header |
| RFC 3515 | SIP REFER |
| RFC 3578 | ISUP Overlap Signaling to SIP |
| RFC 3711 | SRTP (for SIP) |
| RFC 4028 | SIP Session Timer |
| RFC 4568 | SDP Security Descriptions for Media Streams |
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
- GSM-EFR
- 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
Troubleshooting:
- 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
Video:
- 1 DVI output port
AC Power:
- 250W universal for 1U solution
- 350W universal for 2U solution
- DC 400W -48V for 2U (Special Order)
Dimensions:
- 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)
Support and Professional Services:
Sangoma engineers are here to support your success. Whether you need technical support and software maintenance, training, consultation and installation services, Sangoma can help you. Contact your Sales representative for more information
Warranty:
Standard 12-month warranty is included. Additional warranty services available, contact your Sales representative for more information.
Ordering Information
| SKU | Spans | Signalling Links |
| SS7-NSG-AP04 | 4 | 4 |
| SS7-NSG-AP08 | 8 | 8 |
| SS7-NSG-AP16 | 16 | 16 |
| SS7-NSG-AP32 | 32 | 32 |
To become an authorized Empowered by Sangoma channel partner, please visit http://www.sangoma.com/partners.
To purchase now, Contact an Empowered by Sangoma Distributor, Reseller, or Solution Partner near you. Look for the Empowered by Sangoma Logo.
White Papers
SS7 Gateway: Six Good Reasons to Consider Sangoma
Mobile Value-Added Services with Sangoma
Bridging Open Source and Proprietary Telephony Environments
Specialized Hardware Answers Booming VoIP Transcoding Demands
Support Documents
NetBorder SS7 Gateway Use Cases
SIP Network to SS7 Interconnection
Connect a VoIP network to the PSTN, providing connections between SIP phones and conventional and mobile phones.
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