SS7 to VoIP Media Gateway Software

Sangoma’s NetBorder SS7 to VoIP Gateway software provides full-featured, carrier-class VoIP deployments while leveraging the flexibility of standard computing platforms and operating systems.  The software version of this product provides maximum flexibility for developing new or enhanced gateway products.

The NetBorder SS7 to VoIP Gateway allows telecom service providers to introduce VoIP in their networks in the most cost-effective and flexible way. This is simply accomplished by combining the software with Sangoma’s award-winning digital T1/E1 and transcoding boards on standard computing servers. The combination works as a full-fledged SS7 to VoIP gateway, with the flexibility and expandability of software.

The solution supports up to 32 T1/E1 per server. For larger installations (up to 256 T1/E1), distribution across multiple servers provide maximum flexibility to support growth.

Benefits

• Wide range and support of SS7 PSTN protocols and variants
• Scalable
• Flexibility of software deployments – not stuck with monolithic hardware platforms
• Low cost installation leveraging Open Source and off-the-shelf components
• Robust implementation with distribution, failover and redundancy

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

Distributed Architecture for Large Scale Deployments (future)

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

Codec Transcoding

Transcoding is provided by Sangoma voice transcoding boards.  They provide the following codecs:

• 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

Operating System Support:

• 32-bit and 64-bit Linux; CentOS recommended
• Software delivered as binary package or ISO image

Minimum Server Requirements:

• Varies with size of deployment
• Dual-core CPU with 2GB of RAM
• Consult Sangoma Sales for specifics

Contact us for more information about SS7 licenses, SS7 capacity upgrade licenses, and SS7 services.

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To purchase now, Contact an Empowered by Sangoma Distributor, Reseller, or Solution Partner near you. Look for the Empowered by Sangoma Logo. 

White Papers

Mobile Value-Added Services with Sangoma

Bridging Open Source and Proprietary Telephony Environments

Specialized Hardware Answers Booming VoIP Transcoding Demands

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