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PLANET

VIP-2100

Internet Telephony Trunk Gateway

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PLANET
VIP-2100
Internet Telephony Trunk Gateway
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Beschreibung

Details

To give more flexibility, functionalities, and calling capacity in VoIP deployment, PLANET VIP-2100/2400 E1/T1 trunk gateways present an easy, cost-effective solution to amplifying the power of voice over IP (VoIP). The VIP-2000 trunk gateway series supports the packet telephony-based voice interfaces and signaling protocols on the market. They are capable of SIP/H.323 traffic conversion, feature-rich telephony supplementary service, call routing, and IP QoS support in one solution.

PLANET VIP-2100/2400 is not only a VoIP trunk gateway but also a universal VoIP gateway. The VIP-2100 processes the incoming calls in between H.323, SIP and PSTN, with intelligent call routing mechanism. The VIP-2100/2400 is able to route calls between the PBX, the PSTN and the VoIP network to achieve the best combination of cost and quality. PLANET VIP-2000 series can be implemented in the complicated, inhomogeneous telephony service environment, such as SIP + H.323, SIP+H.323 and PSTN enabled network. More than these, the VIP-2100/2400 supports PSTN and VoIP (H.323/SIP) side prepaid and postpaid service. This provides a built-in practical internal AAA service for small deployment and also an external RADIUS interface for ITSP installation.

PLANET VIP-2000 series not only increases more revenues, but also protects the investment in the VoIP service.

Ordering Information
VIP-2100 E1 / T1 Trunk Gateway (1 x E1 / T1)
VIP-2400 E1 / T1 Trunk Gateway (4 x E1 / T1)


Zusatzinformation

Zusatzinformation

Allgemein

Artikelnummer 8658-0392-01
Hersteller PLANET
Lieferzeit 28 Tage

Eigenschaften

Eigenschaften
  • Concurrent SIP/H.323 voice communications
  • ITU-T H.323 v3 and H.450 supplementary service compliance
  • SIP RFC 2543/3261 standard compliance
  • SIP supplemental service - On Hold, Call Transfer support
  • Built-in calling destination and prefix routing for SIP and H.323 P2P calls
  • Mixed SIP, Gatekeeper and P2P voice calls
  • SIP outbound proxy, redirect and register server support
  • SIP/H.323 T.38 fax relay
  • VoIP to VoIP call conversion - SIP to H.323, SIP to SIP, H.323 to H.323
  • Intelligent PSTN call routing and in-trunk hunting
  • External RADIUS Authentication, Authorization and Accounting
  • Behind NAT friendly for SIP calls
  • Inbound and out of band DTMF transmission
  • Built-in IVR & call-flow controller for PSTN / VoIP calls
  • CDR (Call Detail Record) support
  • Built-in internal user authentication for various VoIP applications