VAD VOIP SIP GSM Billing Gateway With Jitter Buffer NTP Network

Delivery Time:3-5 workdays

VAD VOIP SIP GSM Billing Gateway With Jitter Buffer NTP Network

Quick details

Brand name: DBL type: VoIP  Gateway Protocol: SIP & H.323 Support:VPN, Relay Encryption, VoS, Asterisk
 

Description:

16 Ports SIP GSM Gateway GoIP-16 is a 16 SIM Card Broadband Phone Gateway that had been developed by DBL Co. GoIP-16 SIM Card Broadband Phone Gateway is a new product that connect the GSM and the VoIP seamlessly. To GoIP-16 what is installed on the Mobile SIM Card, you can register the GSM telephone on the VoIP Softswitch.SIP and H.323 agreement are built in the GoIP-16 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-16 support multi-device group, it can be easily combined into arbitrary number of channels of Large Gateway Group.

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

The GoIP features embedded SIP and H.323 protocols with flexible setting. The bi-directional password authentication (call authorization) and trust list authentication greatly minimize the risk of charge losses and the flexible routing function can meet special requirements of various call forwarding. In particular, the GoIP gateway supports multi device groups, with flexible setting of large GSM gateway groups with different channel numbers. With its low price, excellent voice quality, and powerful features, the GoIP series gateway is the first choice for system integrators, traffic operators, and softswitch manufacturers.

Key Features

Open Standard VoIP Protocols (IETF SIP V2)

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 Function

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

Retransmit GSM Caller ID to VoIP terminal

Applications:

1.Call Forward

 1.Call Origination refers to a call initiated from the PSTN or cell phone network is terminated using VoIP.

  2.Call Termination refers to a call initiated as a VoIP call is terminated using PSTN or cell phone network.

  3.As shown in the network topology diagram, a VoIP Service Provider is using GoIPs as call origination and termination devices.

   - A call dialed to a GoIP (right hand side) via GSM is first routed via VoIP and then terminated via a VoIP end point or VoIP Service Provider.

   - A VoIP call originated from the left hand side is routed to a GoIP on the right hand side and then is dialed out as a GSM call.

2) Call Back

  1,Call Back is referring to the telecommunications event that occurs when the originator of a   call is immediately called back in a second call as a response.

  2,GoIP could be used to achieve this function alone or as an terminal that is integrated in an existing call back server / platform.

  3,For standalone operation, GoIP receives a call with caller ID information and then rejects the call immediately without answering the call. GoIP then calls back the caller so that he can dial a phone number to make a call. In this case, GoIP must register to a VoIP Service Provider who can offer terminate the call.

  4,In