GoIP-32 SIM Card Broadband Phone Gateway is a new product that connect the GSM    and the VOIP seamlessly. To GoIP-32 what is installed on the Mobile SIM Card, you can   register the GSM telephone on the VoIP soft switch.SIP and H.323 agreement are   built in the GoIP-32 and configured flexible. Caller ID can be seen by using SIP.    Flexible routing can meet the need of all kinds of call forwarding; even more special is    that GoIP-32 support multi-device group, it can be easily combined into arbitrary    number of channels of Large Gateway Group.

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   gateway gsm

          -- Retransmit GSM Caller ID to VoIP terminal

 

Enhanced Features

            -- Dynamic selection of codec

          -- Advanced jitter buffer

          -- Automatic traversal of NAT and firewall

          -- Echo cancellation for Speakerphone

          -- Comfort noise generation (CNG)         

          -- VLAN / Qos

          -- Router   gateway gsm

          -- 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

          -- 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