Syncronized bandwidth Optimizer for VOIP calls

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

Specific Need:

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.

Skills: Asterisk PBX, Software Architecture, VoIP

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About the Employer:
( 0 reviews ) London, United Kingdom

Project ID: #1239546

3 freelancers are bidding on average $267 for this job


you can get some improvmet by using iax trunks in trunking mode. note, u CAN't get 80%(5times) compression of g723 without quality loss. that is TEORETICALY imposible task. u can get near 30% bandwidth compression by More

$250 USD in 3 days
(89 Reviews)

Hello, I have something to discuss. please check pmb.

$250 USD in 10 days
(6 Reviews)

Asterisk itself has this feature

$300 USD in 2 days
(0 Reviews)