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 DHCP IP), receiving calls from server A and sending to gateways ( quintum gareway for example) or E1 cards
3. Number of Server B can be unlimited
4. Number of Gateways/E1 cards per server B can be unlimited
1. Server A to convert all calls in g723.1 codec and server B to receive all calls in g723.1
2. we need some kind of bandwidth compression system( upto 80% than usual SIP calls in g723.1 codec), from Server A to Server B.