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 (quintum gateway 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
7. 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. iax2 trunks in trunking mode. ( May be need to customize IAX2)
B. Open vpn static mode and dynamic mode
8. Asterisk web billing gui for adding gateways.
Adding client , Prefix , dialing plan viewing active calls, billing cdr ,etc.