Asterisk based bandwidth consumption

Technology for reducing bandwidth consumption of VoIP to a great extent (up to 84%). • Uses lossless compression method, thus can be used with mostly used codecs i.e. [url removed, login to view], [url removed, login to view], GSM etc. • Works from NAT - from real IP and/or public IP, static and/or DHCP. • No change in voice quality and route statistics, rather it can increase Average Call Duration (ACD) to 1 minute. • Works as a third party device/software with any Soft Switch. • “Point-To-Multi Point” technology, requires native devices at both end. It works as like tunnel to secure packets and bypass all kind of firewall. • SUPER BANDWIDTH OPTIMIZER uses lossless compression method and do not use codec for providing better compression, rather it compresses transport (RTP). So it works with all commonly used codecs i.e. [url removed, login to view], [url removed, login to view], GSM, AMR etc. • It works with Session Initiation Protocol (SIP Our main goal to minimize the BW in client side with good quality of voice . We 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