voip gsm termination gateway
Details descriptioin
Bridges the GSM and the IP networks by enabling voice communications. It is installed on the Mobile SIM Card, you can register the GSM telephone on the VoIP Softswitch.it can be easily combined into arbitrary number of channels of Large Gateway Group. Low price, perfect sound quality and powerful function make GoIP GSM inevitable choice of system Integrators, traffic Business and soft switch Manufacturers.
Key Features
Open Standard VoIP Protocols (SIP&H.323)
Single or Multiple Server Registrations
Two 10/100 Ethernet for WAN / LAN connections
Peer-to-Peer IP Calls
Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer
Line Echo Cancellation
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
Basic Features
LEDs for Power, Ready, Status, WAN, PC, FXS
Dial in mode or dial out mode only
Call forward from GSM to VoIP and VoIP to GSM
Dial Plan
Retransmit GSM Caller ID to VoIP terminal
8 port goip
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 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
Operating temperature: 10°C to 40°C (50°F to 104°F)
Physical and Environmental
Storage temperature: 0°C to 50°C (32°F to 122°F)
Power: 12 VDC 2A (110V-220V) (AC/DC adapter included)
Warranty: one year
Picture show
New GOIP module will also hot selling...
Price and support
we are manufacturer and can give you factory price. we can give you professional free support services with the help of our technology support engineers.
FAQ explain
What is the communication network can be supported by GoIP?
GSM network
How about our service?
We have professional engineer to help you about all your tech problems anytime,the satisfaction of our customers is our greatest desire.
What is our main express shipment?
DHL,UPS,FEDEX,TNT,EMS(can ship to anywhere all over the world)
How long we're warranty ?
1year or more
During warranty time,if have problem,how to do?
we'll test by teamview or send back,we'll exchange new in free.
How long about our techinical support?
in free,forever