Specifications

1)1 channel voip gsm gateway,goip 1
2)SIP and H.323
3)NAT transversal and route function
4)Vlan and QoS
5)peer to peer

Overview

GoIP GSM Gateway bridges the GSM 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.

 

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
--DTMF: RFC 2833, In-band DTMF, SIP INFO 
 

  

Application:

Saving Cost on International and Long Distance Calls,when you are in the other countries& you have this gateway at your Local office,you could dial & receive phone call from your customers as you are there without Long Distance fee.
Very excellent solution for internal company with branch offices oversea.

 

Example:
peer to peer:it is a new function that you can use our voip without platform,what is more,it is free roaming for international call.you just pay the local call

Model 1:

 

Model 2:

 

Model 3:

 

Model 4:

 

Common function:

1 PSTN to VoIP

 

Description:using the GoIP to connect with VoIP

2 VoIP to PSTN

Description: using the GoIP to connect with PSTN

3 Calling forward

Description:If you are in china but your main business is in Malaysia, you only put a Malaysia SIM card into the GoIP.In this condition,all calling the SIM number can connect your Phone number in China directly.

4 calling back

Description:When you are using the telephone, and you are want to get  preferential from VoIP Phone anytime and anywhere.You just call the SIM card number which in GoIP,the calling number would send to Server by GoIP,then the server receive the calling number and establish the new calling.Then you will receive the new calling,you just accept the calling,Now you are calling with your customer by server. 

 

 

If you have any doubt or want to know further information,please feel free to contact me.