VoIP T1/E1 Trunk Gateway (H323/SIP/PSTN)

VoIP T1/E1 Trunk Gateway (H323/SIP/PSTN) - Снят с производства или поставки
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VoIP T1/E1 Trunk Gateway (H323/SIP/PSTN)
VIP-2100 VIP-2100


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.


Основные характеристики

• 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

Модель VIP-2100 
LAN 2 x 10/100Base-T Ethernet ports
Voice 1 x E1 / T1
Protocol and standards
Call Signaling Control SIP 2.0 RFC2543/(RFC3261, ITUT H.323 v3 and H.450 compliance
Voice codec G.711A/µ-law, G.723.1 (5.3k/6.3k), G.729A
Fax support Automatic voice / FAX detection
H.323 / SIP T.38 fax relay
ECM Support
T.38 during fast connect
VoIP features
Voice processing VAD (Voice Activity Detection)
CNG (Comfort Noise Generation)
G.168 echo cancellation
Configurable audio payload size
Adaptive Jitter Buffer
Silence suppression
Gain control
Voice traffic conversion H.323 to H.323 Call
H.323 to PSTN Call
H.323 to SIP Call
PSTN to H.323 Call
PSTN to SIP Call
SIP to H.323 Call
SIP to PSTN Call
SIP to SIP Call
VoIP to VoIP RTP un-Routed
VoIP to VoIP RTP Routed
DTMF Transmission Transparent
H.245 signal/ Alphanumeric
H.323 Q.931 UUI
RFC 2833
IVR / Call-flow controller Built-in IVR system
Web-based GUI Drag and Drop interface
Full control of call behavior
PSTN / VoIP IVR functions
Support time duration and balance play back
Powerful call information branch
Collected information validation
Interface Console port, TELNET and Web Browser (HTTP/HTTPs)
Front panel LCD display
User account management
Real time monitor
Password Security
SNMP v2 Trap support
AAA(Authentication / Authorization / Accounting) Built-in AAA mechanism, and external RADIUS AAA support
Environmental Temperature: 0~50 degree C (operating)
Humidity: 5 to 95% (non-condensing)
Emission EMI: FCC part 15, CE / PTT: FCC part 68