Deutsch English



Internet Telephony Trunk Gateway

Mehr Ansichten

Internet Telephony Trunk Gateway
Bitte loggen Sie sich ein um Ihren Preis zu sehen


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)




Artikelnummer 8658-0392-01
Hersteller PLANET
Lieferzeit 28 Tage


  • 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