Write some Software Optimizition Voip Traffic

Bandwidth Optimizer for VOIP calls Traffic. Feature Reduce Bandwidth Consumption with excellent technology, for reduce bandwidth consumption of VOIP termination service provider. By using it, can reduce of bandwidth consumption up to 85% maintaining best voice quality is an innovative technology that not only reduces bandwidth consumption but also work as a strong buffer and a expert coordinator of gateway or all type of VOIP device. Available in CD and flash drive… Compatible with any kind of IP Gateways like Addpac, Dinstar, EuroTech gateways, Quintum analog and digital gateways, GOIP, Cisco, Suncom gateways and any standard sip devices. It will also work with any kind of internet connection like DSL, VSAT, WiMax and 3G by its innovative solution of compressing RTP. Project Description: We need some kind of bandwidth Optimizer system ( up to 80-85 % than usual SIP calls )from Server A to Server B. Project Description: Server A = Asterisk server Server B = Asterisk Client server Explanation of scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (quintum gateway for example) or E1 cards. Server A to Server B voice traffic will be encrypted so that voice port blocked bandwidth can be used for termination. We will used. A. iax trunks in trunking mode. B. Open vpn static mode and dynamic mode C. Tnic static and dynamic mode 3. Number of Server B can be unlimited 4. Number of Gateways/E1 cards per server B can be unlimited 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, and that USB flash drive will be delivered to our Server B type client (ther termination provider) Those USB flash drive will be delivered to our Server B type client (there termination provider) A. Any mini Linux distribution exam- puppy Linux, linux mint B. Fedora desktop distribution C. Centos 5.8 or 6 . or Bootable ISO which can be downloaded and burned into CD to run in any computer 6. For server B need portion of asterisk web interface for adding gateways..prefix or so, viewing 1-Ports , 2- Calls , 3-Allerted , 4-Connected , 5- Con Fails , 6- NC , 7 – PDD , 8- ADC , 9- ASR ,10- Tot CallDur , Specific Need: 1. Server A to convert all calls in g723.1 codec and server B to receive all calls in g723.1 2. We need some kind of bandwidth compression system (up to 80-85% than usual SIP calls in g723.1 codec), from Server A to Server B. 3. Server A to Server B voice traffic will be encrypted so that voice port blocked bandwidth can be used for termination. Continue working on project by building up WEB System Management Interface For main Server adding Billing With full feature, and other options like adding gateways, adding client ,.prefix and so on, and viewing the following : 1-Ports , 2- Calls , 3-Allerted , 4-Connected , 5- Con Fails , 6- NC , 7 – PDD , 8- ADC , 9- ASR ,10- Tot CallDur SyncSwitch - a full featured switching and billing platform where customers can add their carriers and centrally manage all of the SBO deployments, control routings most flexibly and intelligently, do complete billing and generate awesome reports 8-Web Interface System Management: Language Management, Multilanguage’s, English, French, Spanish, Chinese, Arabic, Russia . Features of the System: This project platforms will be design ,development, deployment over the server • We will need a complete product deployment training and assistance documents.