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)