Goip Voip GSM Gateway - Original DBL 1 SIM Goip-1 For IP PBX

Posted May 19      Garki I, Abuja (FCT) State, Nigeria
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Application:
The basic use of our GOIP/VOIP as follows:
PSTN->VOIP,
VOIP->PSTN,
Telephone is a common phone. Softswitchis software installed in a computer. It looks upon as a server, Goip looks as client. The scheme needs GoIP,telephone,softswitch,another phone login softswitch through common Gateway device. Then you can PSTN->VOIP and VOIP->PSTN
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Summary
GoIP GSM Gateway bridges the GSM services and the IP networks. It is ideal for VoIP to GSM 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.
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)
Key Features
Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)
Single Server Registrations
SIP peer-to-peer mode
SIP proxy mode
GSM module – 850MHz, 900 MHz, 1800 MHz, 1900MHz
Advanced jitter buffer
VLAN and QoS support
NAT Transversal
Call forward from GSM to VoIP and VoIP to GSM
Password or Trust list protection for dial in mode and dial out
Comprehensive dial Plan
Retransmit GSM Caller ID to VoIP terminal
Dynamic selection of codecs
Echo cancellation for Speakerphone
Comfort noise generation (CNG)
Voice activity detection (VAD)
Firmware upgrade from GUI
Basic Features
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
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
Supported Standards
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
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