GSM VoIP Gateway (goip gateway) for call termination

Models of GSM VoIP Gateway (goip gateway)
Series	1 SIM port	4 SIM ports	8 SIM ports	16 SIM ports	32 SIM ports
External Antenna	GoIP-1	GoIP-4	GoIP-8	GoIP-16	GoIP-32
Internal Antenna	GoIP-1i	GoIP-4i

GSM VoIP Gateway (goip gateway) for call termination, HyberTone Technology

1, 4, 8, 16, 32 ports GSM VoIP Gateway (goip gateway)(External Antenna series)

1, 4 ports GSM VoIP Gateway (goip gateway)(Internal Antenna series)

Specification

Basic Features:

For call termination (VoIP to GSM) and origination (GSM to VoIP) }
Standard SIP & H.323 protocol
GSM: quad-band 850/900/1800/1900MHz

Major Advantages:

Lightweight and Portable
IMEI Changeable
GSM Base Station Optional
Support SIM Bank/ SIM Sever
Manual/ Automatic Selection Operators
Sending and Receiving SMS and USSD (Web Interface)

Free Software/ Sever:

Remote Control Server: Access Interface Remotely
Relay Server: Relay Encryption (Make Terminals Traversal the NAT without STUN and Outbound Proxy)
SMS Server:

Send Bulk SMS
Provide CDR and ASR
Auto Balance and Recharge

SIM Server:

Rotate SIM Cards on Duty
Set GSM Group (Assign several SIMs Per GSM Port)
Set Talk Time per SIM, Set Day of week, Set Time Range
Monitor CDR, ASR, ACD

Key Features

Provide 1, 4, 8, 16 cellular channels for IP-PBX
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
Quad band GSM module: 850MHz, 900 MHz, 1800 MHz, 1900MHz
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

Enhanced Features

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

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 (STUN)- 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/ÃŽÂ¼ law), G.729A/B, G.723.1
DTMF: RFC 2833, In-band DTMF, SIP INFO
Web-base Management
PPP over Ethernet (PPPoE)
PPP Authentication Protocol (PAP)
Internet Control Message Protocol (ICMP)
TFTP Client
Hyper Text Transfer Protocol (HTTP)
Dynamic Host Configuration Protocol (DHCP)
User account authentication using MD5

Contact Information

Lead Sales: Leo
Website: http://www.hybertone.com
Email: [email protected]

Skype: Leohybertone
Tel: +86 18576773508

http://www.facebook.com/HyberTone.VoIP
http://www.youtube.com/HyberTone