VOIP BANDWIDTH OPTIMIZATION FOR TERMINATION

Project Description: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with SIP and IAX protocol, using G711, G729 and G723.1 codecs transferring / sending incoming calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP / unknown IP), receiving calls from server A and sending to gateways (i.e Goip Gateways) or E1 cards 3. Number of Server B can be multiple / unlimited. 4. Number of Gateways/E1 cards per server B can be unlimited. 5. For server B installation may an ISO of USB image mounting of easy with some free tools which can be done using Puppy Linux OS. 6. For For configuration of both A and B there will some GUI (i.e. Web Interface) for adding gateways prefixes, viewing active calls, billing cdr, etc. Specific Need: 1. Server A to convert all calls in g723.1 codec and server B to receive all calls in g723.1 and forward them to gateways g729 codec. 2. Some kind of bandwidth compression is required for better performance on low bandwidth connection. i.e. TrafficSqeezer. 3. server A to Server B voice traffic will be encrypted so that voice port blocked bandwidth can be used for termination OR you can develop this using wrt or openwrt for tp link or any router