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GOIP32

Update: 2019/10/28      View:
  • Brand:    YX
  • Type:    GOIP32
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Introduction
 

Overview

Description:

The GoIP series gateway is a broadband relay gateway newly developed by YX Technology. It is a new product for seamless connection between the GSM network and VoIP network. When the mobile phone SIM card is installed in the GoIP, users can register the GSM phone to the VoIP softswitch system. Through the GoIP, users can realize the uplink and downlink calls between the GSM network and the VoIP network. In addition, the GoIP supports the transparent transmission of the caller number from the PSTN to the VoIP.

GoIP is designed to work in conjunction with key phone systems and IP-PBX to provide GSM communications. The extensive compatibility of the GoIP makes it an ideal choice to be deployed in multi-vendor open architecture VoIP networks. GoIP is a great way to provide fast phone service deployment where regular PSTN line may not be readily available. GSM gateway also provides significant savings in usage, infrastructure and maintenance cost compared to conventional PSTN.

The GoIP features embedded SIP and H.323 protocols with flexible setting. The bi-directional password authentication (call authorization) and trust list authentication greatly minimize the risk of charge losses and the flexible routing function can meet special requirements of various call forwarding. In particular, the GoIP gateway supports multi device groups, with flexible setting of large GSM gateway groups with different channel numbers. With its low price, excellent voice quality, and powerful features, the GoIP series gateway is the first choice for system integrators, traffic operators, and softswitch manufacturers.


Basic Function 

● LEDs for Power, Ready, Status, WAN, PC, GSM 
● Call forward from GSM to VoIP and VoIP to GSM
 
● Dial in mode or dial out mode only
 
● Dial Plan
● Password protection for both GSM dial in or dial out

● Retransmit GSM Caller ID to VoIP terminal

Features

Key Features:
● Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)
● Single or Multiple Server Registrations
● Two 10/100 Ethernet for WAN / LAN connections
● GSM module for making GSM calls
● Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer
● VLAN and QoS support
● NAT Transversal and Router functions
● Voice prompts, HTTP Web, Auto Provision support for configuration and updates
● Highly stable embedded Linux operating system in high performance ARM 9 Processor

Enhanced Features
● Dynamic selection of codec
● Advanced jitter buffer
● Automatic traversal of NAT and firewall
● VLAN / Qos
● Router
● Echo cancellation for Speakerphone
● Comfort noise generation (CNG)
● Voice activity detection (VAD)
● Auto provisioning (requires auto provisioning server)
● On line firmware upgrade
● Multi-language support: English and Chinese

Advantages

Competitive Advantage: 

● Remote SIM operation for SIM Card management

● SMPP support for 3rd party development of SMS Applications.

● Free server utilities for remote access and SMS management.

● Telnet mode for sending AT commands to GSM module.

● Compact and light weight design

● Direct customer support

Physical Features

 Hardware

 Parameters

 Remark

 Model

 GoIP-32

 

 Processor

 ARM11 700MHZ

 

 DSP

 VP-101-1 196MHz

 

 RAM

 128M

 

 FLASH

 8M

 

 Network Card

 100/10BASE-T ×2

 

 LED

 operating and circuit light

 

 Power consumption

 12V/DC /4.5A +-10%

 

 Weight

 5kg

 Contains the antenna

 Size

 37.5*22*17(cm)

 

 Operating Temperature

 045

 

 Working Humidity

 40%-90 non-condensation

 

 Color

 grey

 

 SIM port

 32

 

 RJ45 port

 1

 

 Speech Characteristics

 Service Condition

 Remark

 G.168 Echo cancellation

 Support

 16mS

 g.723

 Support

 

 g.711A/u

 Support

 

 g.729A/B

 Support

 

 GSM

 Support

 

 PLC

 Support

 

 CNG

 Support

 

 VAD

 Support

 

 Jitter Buffer

 Support

 

 T.38

 Nonsupport

 

 Network Characteristic

 Parameters

 Remark

 LAN Port

 DHCP

 Support

 

 PPPoE

 Support

 

 Static IP

 Support

 

 NAT Transversal

 Relay or Port forwarding

 Relay need coordinate with Relay Server

 Network time

 NTP / SNTP

 

 10/100BASE-T

 10/100 auto adaptation

 

 PC Port

 Static IP

 Support

 

 DHCP

 Max support 200 terminals

 

 10/100BASE-T

 10/100 auto adaptation

 

 Switch mode

 Support

 

 Protocols

 ITU H.323 V4 and IETF SIP V2, SIP (RFC3261) TCP/UDP/IP, RTP/RTCP, HTTP, ARP/RARP, 

ICMP, DNS, DHCP, NTP, TFTP, TELNET, PPPoE

 Support

 User Interface

 Parameters

 Remark

 Web page Configuration

 HTTP

 

 Multiple password

 3 password

 Customized service

 Keyboard setup

 Support

 

 Online update

 http

 

 Broadcast IP

 Support

 Chinese , English

 Billing

 Support

 Coordinate with YX Billing Software

 Language

 Chinese, English

 

 Multi-regional warning tone

 China,HK,USA,UK,German etc.

 

 Warranty

 One year

 

Applications

1. User Manuals
     GoIP Series User Manual V1.4
     GoIP Series User Manual V1.0
     The lastest Firmware Version

2. Call Forward 
1.Call Origination refers to a call initiated from the PSTN or cell phone network is terminated using VoIP. 
2.Call Termination refers to a call initiated as a VoIP call is terminated using PSTN or cell phone network. 
3.As shown in the network topology diagram, a VoIP Service Provider is using GoIPs as call origination and termination devices. 
    - A call dialed to a GoIP (right hand side) via GSM is first routed via VoIP and then terminated via a VoIP end point or VoIP Service Provider. 
    - A VoIP call originated from the left hand side is routed to a GoIP on the right hand side and then is dialed out as a GSM call. 

3. IP PBX Call Origination and Termination 
1.Instead of FXO gateways, GoIP are as a call termination and origination device for the IP PBX as shown in the diagram above.
2.VoIP endpoints connected to the IP PBX can make calls to cellular/traditional telephone network via the GoIP GSM ports.
3.Outside callers can then call in via the GoIP GSM ports to reach any of the VoIP endpoints that are registered to the IP PBX. 
4.GoIP can be configured in a group mode such that all GSM ports can be used by just dialing only one GSM number. Please refer to the Call Center Application for more information. 

4. Sending Bulk SMS Service 
1.Sending bulk sms text messages is a common technique for telemarketing to reach the target customers.
2.A bulk SMS system can be implemented quickly and easily using GoIPs and our proprietary SMS server. Telemarketers are now have full control on how and when they want to send text messages.
3.In addition, SMS text messages are now used widely in many companies, organizations, schools, clubs as a mean for broadcasting information. They can now build their own SMS system without paying expensive charges to their GSM server provider.
4.This system can also take the advantage of using the same GSM service provider to send sms to the phone subscribers in the same service provider.

5. Call Back
1,Call Back is referring to the telecommunications event that occurs when the originator of a call is immediately called back in a second call as a response.
2,GoIP could be used to achieve this function alone or as an terminal that is integrated in an existing call back server / platform. 3,For standalone operation, GoIP receives a call with caller ID information and then rejects the call immediately without answering the call. GoIP then calls back the caller so that he can dial a phone number to make a call. In this case, GoIP must register to a VoIP Service Provider who can offer terminate the call.
3,In a call back system, GoIP acts as a device to initiate the call back function. Typically, this is done in two ways. The first method is to send an SMS with the callee’s phone number to the GoIP. The GoIP then sends both the caller’s and callee’s phone numbers to the call back server to complete the call back function. The second method is to call the GoIP and the hang up (with the call being answered). GoIP sends the caller’s phone number to the call back server and the call back server calls the caller directly so that the caller can then dial a phone number to make a call.


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