bandwith optimizer and security - open to bidding

Project Description: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with SIP and IAX protocol, using G711, G729 and G723.1 codecs transferring / sending incoming calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP / unknown IP), receiving calls from server A and sending to gateways (i.e Goip Gateways) or E1 cards 3. Number of Server B can be multiple / unlimited. 4. Number of Gateways/E1 cards per server B can be unlimited. 5. For server B installation may an ISO of USB image mounting of easy with some free tools which can be done using Puppy Linux OS. 6. For For configuration of both A and B there will some GUI (i.e. Web Interface) for adding gateways prefixes, viewing active calls, billing cdr, etc. Specific Need: 1. Server A to convert all calls in g723.1 codec and server B to receive all calls in g723.1 and forward them to gateways g729 codec. 2. Some kind of bandwidth compression is required for better performance on low bandwidth connection. i.e. Traffic Squeezer. 3. server A to Server B voice traffic will be encrypted so that voice port blocked bandwidth can be used for termination OR you can develop this using wrt or openwrt for tp link or any router 1 ) We need Complete GUI and Web interface to manage Client and admin Side , 2) also Should be added option for changing ip adress incase of Shifting server when ip get blocked ,Skills required: Asterisk PBX, MySQL, PHP, Telecommunications Engineering, VoIP Project Description: PLEASE DON'T BID IF YOU DON'T KNOW WHAT IT IS ABOUT send query before bidding Server X = Asterisk Server Server Y = Window based Asterisk Client server Main goal minimize Bandwidth in client side with quality voice . Required bandwidth compression (upto 60-80% in reality then theory) from Server X to Server Y. A usual SIP in G729 call takes 27-32kbps per port Explanation of scenario: 1. Server X (asterisk server, with static IP) receiving VoIP calls from different Carrier/Originator, with h323/sip protocol, using G711,G729 and/or G723r6.3 codec and sending calls to Server Y. 2. Server Y {Windows based Asterisk server with PRIVATE NETWORK IP, receiving calls from server X and sending to gateways (quintum, goip etc brand) or E1 cards. (Please take a look at this page for better clarity [login to view URL]) 3. Number of Server Y can be unlimited. 4. Number of Gateways/E1 cards per server Y can be unlimited 5. For server Y installation will be window based software based on Asterisk can be easily downloaded from our website and can be installed must be in [login to view URL] file mode ([login to view URL] for reference). This installation file will require specific activation code from us to activate otherwise will not run. And each time this file run must register with our Server X to see if the activation code has been expired or not. We want our client side to be based on windows not something on USB flash dirve linux. 6. Server X must have codec conversion i.e., can change the voice codec of SIP G729 to SIP G723r63 or visa verse 7. Server X to Server Y voice traffic will be encrypted so that voice port blocked bandwidth can be used for termination this is very important as this service will mainly be used for voip block countries. we will use: - iax trunks in trunking mode. - Open vpn static mode and dynamic mode - Tnic static and dynamic mode 8. We need - Commercial class Asterisk web billing - GUI for adding gateways - Adding client - Setting Dialing plan /Prefix - viewing active calls, - Real time billing cdr so that can be use to bill the call originator - Localization of the cdr as per time frame of the originator to match the billing of the originator - Real Time ASR, ACD per Gateway, Per IP, Per destination based on 15 minutes/ 30 minutes/ 45 minutes/ 60 minutes, daily, weekly and monthly basis can be used set standard to block call below a certain level of ASR and ACD. To satisfy the car