Asterisk VoIP Call Bandwidth Optimize (urgent)

any one can please give me this solution?

or if u want to sell ur solution please contact me at

skype : lazy.mind1

i need it very urgent.
( i m a student in computer science . only passed 6 month)
so if u help me it will be great full for me. or if u want sell also tell me. again i m a student from bangladesh so don’t tell any higher amount. i can’t bear it.

if u want to make deal . also welcome :smile:

I need some kind of bandwidth compression system ( upto 60-80% than usual SIP calls )from Server A to Server B.

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 (addpac, dinstart gateway for example)

  3. Number of Server B can be unlimited.

  4. For server B installation need easy to use ISO image that could be booted from USB flash drive, and those USB flash drive will be delivered to our Server B type client (ther termination provider)
    A. Any mini Linux distribution exam- puppy Linux , linux mint, Centos 5.8 or 6

  5. 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( optional )

  6. Asterisk web billing gui for adding gateways.
    Adding client , Prefix , dialing plan viewing active calls, billing cdr ,etc.

here I add some company we need similar thing