| VG800 IP Voice Gateway
 The
VG800 IP Voice Gateway is manufactured by Maipu, China second
largest router manufacturer. The gateway supports both outbound
voice and fax transmission. It allows up to eight simultaneous
outbound calls to be made from eight different traditional
phones while sharing a single broadband connection.
To place a call, the user simply enters a PIN and then the
destination phone number. The PIN is universal and can be
used from any phone connected to a VG800 gateway anywhere
in the world. The calls are charged at deeply discounted long
distance rates.
Another important feature of the VG800 is its ability to
accept regular phone lines as input and serve them as direct
inward dial (DID) numbers. This also allows local DID’s
to be available from regions where DID’s are difficult
to obtain. Callers can then dial the DID number from any traditional
or mobile phone and then the destination number to take advantage
of the deep discount long distance rates.
Call shops and outbound call centers that use traditional
phone equipment can significantly reduce their operating cost
by using the VG800 and allow phone calls to be made via the
Internet. The shops’ revenues will derive from selling
PIN cards. Their customers can either make calls on the premises
by using the phones connected to the shop’s gateway
or from a remote or mobile phone by calling the gateway’s
DID number. The gateway can also be configured for outbound
call centers so that no PIN is required when placing calls.
Features:
* Modularized Structure Design
* Flexible Access
* On-demand Investment
* Auto-switching
* Guaranteed QOS
* Fine Compatibility
* Manageability
* Low cost
* Security
Technology specifications:
System Features
* Processor Motorola MPC860T based 50M Hz
* Memory 64Mbyte
* flash 8Mbyte
* 10M/100M Ethernet 1 or 2, support for full/half duplex auto-negotiation
* Configuration interface(console) 1, RJ45 interface,asynchronous
DTE working mode
Telephone features
* PSTN Gateway FXS, FXO
* User terminal POTS phone ,fax, ePhone, NetMeeting client
* Call type Analog user to analog user, analog user to IP
user, analog trunk to IP user
* PBX characteristic PBX functions such as call transfer,
caller-id etc.
* Echo Cancellation 25mSec,G.168 compliant
* Voice activate/Comfortable noise VAD/CNG
* Signaling check and generation DTMF
System interfaces
* IP Network-side interface 1 10/100Base-T, 1 10M Ethernet
port to configure
* Analog line (Z interface)interface Maximal number of 16
FXS/FXO ports to configure
Protocols
* Ethernet interface IEEE 802.2 (LLC), IEEE 802.3 (Ethernet)
* E1 interface ITU-T G.703 (1999)
* Relay protocol Analog - CO Trunk
* Routing protocol Static protocol?RIP v1 and V2 (RFC 2453)?
OSPF?IRMP
* VPN and firewall IPSec (IPESP), MD5, SHA-1, DES, Triple-DES,
IKE (ISAKMP), NAT
* VoIP signaling protocol ITU-T H.323, H.225.0 (RAS and Q.931),
H.245, H.450.x
* Voice coding G.711?G.723?G.729
* Voice compression transmission RTP/RTCP
* Management protocol SNMP
Legal Standards
* Electromagnetic interference/ Electromagnetic compatibility
FCC Part 15, Class B, EN55022 and EN50082-2 (See Note 2)
* Static release IEC1000-4-2 and GR-1089-CORE
System Performance
* Average call building time <4s(China NO.1)
* System transmission delay (multiple) <145ms(G.729A);
<185ms(G.723.1)
* Multiple-call completing rate >90%
* Long-time call holding rate >99%
* DTMF signal frequency offset <1.2%
Power consumption
* Working temperature 0oC-+40oC
* Humidity 10-90% non-condensing
Working Environment
* Working voltage(AC) 90~250V
* Power(W) 50W |
