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)