gateway for call termination

 VoIP GSM Gateway GoIP  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)

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

 gateway for call termination

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

 gateway for call termination

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

 gateway for call termination

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

Operating temperature: 10°C to 40°C (50°F to 104°F)

 gateway for call termination

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