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The GoIP series gateway is a broadband relay gateway 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
GoIP GSM Gateway bridges the GSM services and the IP networks. It is ideal for VoIP to wireless services where a fixed telephone line (PSTN) is not available or for cell phone roaming via the VoIP network.
GoIP GSM Gateway 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 GSM Gateway is a great way to provide fast phone service deployment where regular PSTN line may not be readily available. GoIP GSM gateway also provides significant savings in usage (long distance or international), infrastructure and maintenance cost compared to conventional PSTN.
1. 4 GSM channels,up to 4 sim cards
2. Quad band ,IMEI changeable
3. Support of SMB32 SIM Bank
4. VoIP SIP,H323,Remote Access
5. Optional SMS termination
6. Easy to install and administrate
7. Auto Balance and Recharge
Benefits of GoIP:
Extensive product compatibility with industry leading vendor
Cost-Savings on phone calls between mobiles or to PSTN
Easy to install – IP device with Web based management interface
Can be managed and monitored remotely over Internet - a great service to offer to customers by system integrators
GoIPs can be grouped together to establish GSM gateway cluster
Termination between GSM/VoIP
Schedule or on-demand SMS Broadcast messages to users
(Additional SMS server is required)
Multiple GoIP grouping mode
Provide one cellular channels for IP-PBX
Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)
Single or Multiple Server Registrations
Two 10/100 Ethernet circuits connect to the LAN and an additional device
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
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
Password protection for both GSM dial in or dial out
Retransmit GSM Caller ID to VoIP terminal
Dynamic selection of codec
Advanced jitter buffer
Automatic traversal of NAT and firewall
VLAN / Qos
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
ITU: H.323 V4, H.225, H.235, H.245, H.450
RFC 1889 - RTP/RTCP
RFC 2327 SDP
RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
RFC 2976 SIP INFO Method
RFC 3261 SIP
RFC 3264 Offer/Answer model with SDP
RFC 3515 SIP REFER Method
RFC 3842 A Message Summary and Message Waiting Indicator
RFC 3489 Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)
RFC 3891 SIP Replaces Header
RFC 3892 SIP Referred-By Mechanism
draft-ietf-sipping-cc-transfer-04 Session Initiation Protocol Call Control - Transfer
Codec: G.711 (A/µ law), G.729A/B, G.723.1
DTMF: RFC 2833, In-band DTMF, SIP INFO
Hardware Specifications:Hardware Parameters Remark Model GoIP-4 Processor ARM9 133MHZ DSP VP-101-1 196MHz RAM 16M FLASH 4M Network Card 100/10BASE-T ×2 LED operating and circuit light Power consumption 12V/DC Weight 0.65kg Size 32.5*14*6(cm) Operating Temperature 0－45℃ Working Humidity 40％－90％ non-condensation Color grey SIM port 8 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 DBL 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,
In order to promote our product ,Now we have added some new funcitions in GoIP.By adding the new funcitions,you can build the call without using the softwhich or platform,just depending on the internet.it makes our calls more convenient and easier.what’s more, it can save a lot of call fee.Here is the following example:
peer to peer:it is a new function that you can use our voip without platform,what is more,it is free roaming for international call.you just pay the local call
1. Call Forward
a. Call Origination refers to a call initiated from the PSTN or cell phone network is terminated using VoIP.
b. Call Termination refers to a call initiated as a VoIP call is terminated using PSTN or cell phone network.
c. 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.
2. IP PBX Call Origination and Termination
a. Instead of FXO gateways, GoIP are as a call termination and origination device for the IP PBX as shown in the diagram above.
b. VoIP endpoints connected to the IP PBX can make calls to cellular/traditional telephone network via the GoIP GSM ports.
c. 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.
d. 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.
1 * 4 Channel GSM VOIP Gateway GOIP 4
1 * Power Adapter
1 * Ethernet Cable
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This item is covered by 1 Year Return To Base Warranty
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In the unlikely event that change of mind and a return is request,you can return it back within 14 days since when you receive it. All returns must include all original items undamaged, in re-saleable condition, all original packaging.
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