Abc project for voip softswitch in database

Project Description: 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 carrier to fulfill a certain level of quality assurance. - Must able to generate error 503 for SIP Protocol when the termination side is down to assimilate the call generation side to stop sending the call - Must have multi-user login so that user of different client of server Y can log in and set their own parameters such as client, codec, destination, origination, protocol, gateways, dialplan, price, costing, billing etc - we will provide Dedicated server asterisk and client asterisk - Require to configure IAX trunking, BW compression can be measured making the SIP-> IAX call trunking, need a simple WEB tool to change IAX IP and port (you understand that it is sensitive option when trunk is blocked by country border GW) - Install OpenVPN Server&client - at this stage we will test it and measure the BW compression with all kinds of options like codecs and openvpn compression modes; - require continue project with compiling the automated installation distribution (with OpenVPN, Asterisk, Codec conversion, IAX trunks config ) for client-side CentOS system, which can be distributed to may servers. - continue working on project by building up WEB interface for main server adding Billing, adding IAX trunks and other options for user friendly GUI - WE REQUIRED THE SAME SERVICE AS [login to view URL] AND/OR [login to view URL] - MUST BE ABLE TO WORK Successfully with internet service provider in Bangladesh and UAE -- Banglalion -- Qubee -- Teletalk 3G -- ollo usb doungle