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VIP-450
Port FXO SIP Internet Telephony Gateway


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To meet the next generation Internet telephony service demands, PLANET VIP-450FO delivers next-generation platform for voice services and applications, the VIP-450FO is reliable, efficient and TOC-saving; widely interoperable; and scalable and manageable solution for your VoIP network. The VIP-450FO is an entirely new kind of voice delivering superlative functionality and stellar audio quality.

PLANET VIP series are designed for comfort, ease-of-use with a sophisticated, and satisfaction from customers, The VIP-450FO not only inherits traditions of quality voice communications and real-time fax data over IP networks, but also eliminates the human resource VoIP network deployment. With optimized SIP architecture, PLANET VIP-450FO is the ideal choices for P2P voice chat, ITSP cost-saving solution, but also provide network-converting feature to translate the packet network into traditional PBX system.

With built-in PPPoE/DHCP/DDNS clients, up to 4 concurrent connections in the VIP-450FO, voice communications can be established from anywhere around the world. PLANET VIP-450FO comes with intuitive user-friendly, yet powerful management interface (web/telnet/console), that can dramatically reduce IT personnel resource, and complete VoIP deployment in a short time, plus remote management capability, VoIP administrators can monitor machine/network status, or proceed maintenance/trouble-shooting service via Internet browser or telnet session.

With built-in PPPoE/DHCP/DDNS clients, up to 4 concurrent connections in the VIP-450FO, voice communications can be established from anywhere around the world. PLANET VIP-450FO comes with intuitive user-friendly, yet powerful management interface (web/telnet/console), that can dramatically reduce IT personnel resource, and complete VoIP deployment in a short time, plus remote management capability, VoIP administrators can monitor machine/network status, or proceed maintenance/trouble-shooting service via Internet browser or telnet session.
Key Features
Standards compliant & Surpassing voice quality
PLANET VIP-450FO is SIP version 2 compliant; more than this, it provides superior PBX extension/ PSTN lines connectivity without changing user's that are interoperable with major SIP gateways/agent/proxy servers. Retaining excellent traditions in PLANET VoIP products, the VIP-450FO prioritizes voice packets using IP precedence, and combine state-of-the-art technology of voice packet handling, including echo, noise reduction, voice reconstruction and redundancy to provide customers toll quality VoIP communication.
VoIP and Network conversion
Via configurable voice codec: G.723.1 , G.729ab and G.711, the VIP-450FO supports for multiple algorithms to meet different VoIP application demands. The VIP-450FO supports converting telephony protocols (SS7, Analog (FXS/FXO)) into packet switching network between calling and called voice gateways or PBX system, in a heterogeneous signal-switching environment.
Domain name call & DDNS supports
Either IP or URL addressing, the VIP-450FO is able to find, and communicate with destination SIP gateway/agents. Meanwhile, DDNS service is very useful to those VoIP gateways deployed in a dynamic IP environment. Collaborate connection agents (PPPoE, or DHCP clients), built-in DDNS client in the VIP-450FO can help those who have no static IP address to map dynamic IP address to an easy-to-remember URL, and this will allows other 3rd party SIP compatible voice gateways/agents to allocate position of VIP much easier than ever. (To establish voice communication via domain name, please make sure the other party gateway/agents supports domain name call.)
SIP registration, SIP authentication
The VIP-450FO is able to perform SIP registration, authentication, to interact with major SIP gateway/IP Phone on the market.
SPECIFICATION

Model

VIP-450FO

Ports

LAN

1(10Base-T/100Base-TX, Auto-Negotiation)

Voice

4 (4 x FXO)
sip Specification
Call Signaling Control SIP 2.0 (RFC3261) / SIP over UDP (IETF RFC2543)
Simultaneous connection Up to 4 channels voice
Voice processing Voice Active Detection, DTMF detection/ generation, G.168 echo cancellation (16mSec.), Comfort noise generation, Call progress detection, Gain Control
Internet sharing
Protocol
TCP/IP, HTTP, DNS, RTP, RTCP, SDP
Management RS-232 Console /Telnet, HTTP
Hardware Specification
LED Indicators System: 2, PWR, CPU
LAN: 4, 100, 10, LNK, COL
Voice: 4 In-Use/Ringing
Power Requirement 12V DC
Dimension (L x W x H mm) 145 x 240 X 44 (Metal Case)
Environmental Temperature: 0~50 degree C (operating)
Humidity: up to 90% (non-condensing)
Emission EMI: FCC part 15, CE / PTT: FCC part 68

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SEITEL Telecomunicações Ltda - Cx. Post. 06 - 95860-000 - Taquari / RS / Brasil - Fone: +55 (51) 3653-1254 / Fax: (51) 3653-1844