The basic use of our GOIP/VOIP as follows:

PSTN->VOIP,

VOIP->PSTN,

Telephone is a common phone. Softswitchis software installed in a computer. It looks upon as a server, Goip looks as client. The scheme needs GoIP,telephone,softswitch,another phone login softswitch through common Gateway device. Then you can PSTN->VOIP and VOIP->PSTN

GoIP GSM Gateway bridges the GSM services and the IP networks. It is ideal for VoIP to GSM services where a fixed telephone line (PSTN) is not available or for cell phone roaming via the VoIP network.

GoIP GSM Gateway is designed to work in conjunction with key phone systems and IP-PBX to provide GSM communications. The extensive compatibility of the GoIP makes it an ideal choice to be deployed in multi-vendor open architecture VoIP networks.

GoIP GSM Gateway is a great way to provide fast phone service deployment where regular PSTN line may not be readily available. GoIP GSM gateway also provides significant savings in usage (long distance or international), infrastructure and maintenance cost compared to conventional PSTN.

Extensive product compatibility with industry leading vendor

Cost-Savings on phone calls between mobiles or to PSTN

Easy to install – IP device with Web based management interface

Can be managed and monitored remotely over Internet – a great service to offer to customers by system integrators

GoIPs can be grouped together to establish GSM gateway cluster

Termination between GSM/VoIP

Schedule or on-demand SMS Broadcast messages to users

(Additional SMS server is required)

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

Single Server Registrations

SIP peer-to-peer mode

SIP proxy mode

GSM module – 850MHz, 900 MHz, 1800 MHz, 1900MHz

Advanced jitter buffer

VLAN and QoS support

NAT Transversal

Call forward from GSM to VoIP and VoIP to GSM

Password or Trust list protection for dial in mode and dial out

Comprehensive dial Plan

Retransmit GSM Caller ID to VoIP terminal

Dynamic selection of codecs

Echo cancellation for Speakerphone

Comfort noise generation (CNG)

Voice activity detection (VAD)

Firmware upgrade from GUI

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

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

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

In order to promote our product ,Now we have added some new funcitions in GoIP.By adding the new funcitions,you can build the call without using the softwhich or platform,just depending on the internet.it makes our calls more convenient and easier.what’s more, it can save a lot of call fee.Here is the following 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