ETF SIP V2 GSM FXS Gateway

Quick detail:

• Brand name: DBL
• Model Number: GS-1I
• type: VoIP  Gateway
• Ports: 1-GSM&1-FXS
• Size: 13*10*3.5cm
• Protocol: SIP & H.323
• Support: voip, gsm
   

Description:

GS-1 is a GoIP-1 with one FXS port VoIP Gateway , it bridges the GSM,Analog telephone,and the IP networks by enabling voice communications. It is ideal for VoIP to Local termination where a fixed telephone line (PSTN) is not available or for cellphone roaming via the a VoIP network. Significant savings on long distance charges can be realized. Low price, perfect sound quality and powerful function make GoIP GSM inevitable choice of system Integrators, traffic Business and soft switch Manufacturers.

Application

1. From VoIP to PSTN

2, PSTNàGoIPàPSTN( without long distance charge.)

Competitive Advantage:

Key Features

Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)

Single or Multiple Server Registrations

Two 10/100 Ethernet circuits connect to the LAN and an additional device

GSM module for making GSM calls

Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer

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, GSM

Call forward from GSM to VoIP and VoIP to GSM

Dial in mode or dial out mode only

Dial Plan

Password protection for both GSM dial in or dial out

Retransmit GSM Caller ID to VoIP terminal

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

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