VoIP T1/E1 Trunk Gateway (H323/SIP/PSTN)
To give more flexibility, functionalities, and calling capacity in VoIP deployment, the PLANET VIP-2100/2400 E1/T1 trunk gateways present an easy, cost-effective solution to amplify the power of Voice over IP (VoIP). The VIP-2000 series trunk gateways support 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 are 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 are 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 trunk gateways 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, VIP-2100/2400 support 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 trunk gateways not only increase more revenues, but also protect the investment in the VoIP service.
• 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
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