Bandwidth optimization voip solution - open to bidding

we need basically bandwith optimization. in client end they have termination gateways, each gateway can carry 30 simultaneous calls. and all gateway under DHCP network . we take a server which run in static IP . all client should send calls to server IP from his SIP server and we need to create separate account for each gateway . and calls should send to specific termination . call should pass with sip / iax2 , g729 or g723. in client end we want to setup asterisk module which can convert calls from sip / IAx to sip and pass the calls to gateway . 01. asterisk or SBO server. which receive calls from many sip server. 02. asterisk./SBO transfer calls to local PC or router . 03. in router / pc have a module which can route calls to LAN IP . 04. in LAN IP there have a termination gateway . so calls can pass normal . the thing is , u need to work on asterisk/ SBO server and client end PC/ router. for your understand . here i give you a web site . as they provide .. we need same solution, [login to view URL] check the site . we need same solution. waiting for your update. regards, Md. Abu NomanAdditional Project Description: 06/20/2012 at 2:51 EDT 1 # VOIP Call Termination Over Local IP ( VOIP Over NAT Traversal Or Firewall ) 2 # VOIP Call & Quality Voice Termination In Lower Bandwidth ( 1 Call 6-8kb ) 3 # Web Based Administrator, Agent & Client Control ( Easy User Interface) 4 # Easy New Gateway Add System For Client ( Add Gateway:- PC MAC Address & Gateway Local IP ) 5 # Stay Alive Internet System ( As Like Skype ) 6 # Easy Asterisk Billing System 7 # Easy Bootable USB Client 8 # VOIP Server Connect The Common Configuration Client Over Local IP & MAC Address. 06/20/2012 at 2:53 EDT well , we need basically bandwith optimization. in client end they have termination gateways, each gateway can carry 30 simultaneous calls. and all gateway under DHCP network . we take a server which run in static IP . all client should send calls to server IP from his SIP server and we need to create separate account for each gateway . and calls should send to specific termination . call should pass with sip / iax2 , g729 or g723. in client end we want to setup asterisk module which can convert calls from sip / IAx to sip and pass the calls to gateway . 01. asterisk or SBO server. which receive calls from many sip server. 02. asterisk./SBO transfer calls to local PC or router . 03. in router / pc have a module which can route calls to LAN IP . 04. in LAN IP there have a termination gateway . so calls can pass normal 05. VOIP Call Termination Over Local IP ( VOIP Over NAT Traversal Or Firewall ) 06. VOIP Call & Quality Voice Termination In Lower Bandwidth ( 1 Call 6-8kb ) 07. Web Based Administrator, Agent & Client Control ( Easy User Interface) 08. Easy New Gateway Add System For Client ( Add Gateway:- PC MAC Address & Gateway Local IP ) 09. Stay Alive Internet System ( As Like Skype ) 10. Easy Asterisk Billing System 11. Easy Bootable USB Client 12. VOIP Server Connect The Common Configuration Client Over Local IP & MAC Address. . the thing is , u need to work on asterisk/ SBO server and client end PC/ router.