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VIP-2100

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Togivemoreflexibility

詳細介紹

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.

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 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

端口
LAN 端口	2 x 10/100Base-T Ethernet ports
語音端口	1 x E1 / T1
協議與標準
呼叫信號控制	SIP 2.0 RFC2543/(RFC3261, ITUT H.323 v3 and H.450 compliance
語音編碼	G.711A/?-law, G.723.1 (5.3k/6.3k), G.729A
支持	Automatic voice / detection H.323 / SIP T.38 relay ECM Support T.38 during fast connect
VoIP功能
語音處理技術	VAD (Voice Activity Detection) CNG (Comfort Noise Generation) G.168 echo cancellation Configurable audio payload size Adaptive Jitter Buffer Silence suppression Gain control
語音流量轉換技術	H.323 to H.323 Call H.323 to PSTN Call H.323 to SIP Call PSTN to H.323 Call PSTN to PSTN 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 傳輸	Transparent H.245 signal/ Alphanumeric H.323 Q.931 UUI RFC 2833 SIP INFO
IVR/呼叫流程控制	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
管理
接口	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（認證//計費）	Built-in AAA mechanism, and external RADIUS AAA support
環境
工作溫度&濕度	Temperature: 0~50 degrees C Humidity: 5 to 95% (non-condensing)
認證	EMI: FCC part 15, CE / PTT: FCC part 68