Our main goal to minimize the BW in client side with good quality of voice .


We 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

5. For server B installation need easy to use ISO image that could be booted from USB flash drive, and those USB flash drive will be delivered to our Server B type client (ther termination provider)

A. Any mini Linux distribution exam- puppy Linux , linux mint

B. Fedora desktop distribution

C. Centos 5.8 or 6

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. iax trunks in trunking mode.

B. Open vpn static mode and dynamic mode

C. Tnic static and dynamic mode

8. Asterisk web billing gui for adding gateways.

Adding client , Prefix , dialing plan viewing active calls, billing cdr ,etc.

we will provide you the Dedicated server asterisk and client asterisk

configure IAX trunking, so we can measure the BW compression making the SIP-> IAX call trunking, need develop a simple WEB tool to change IAX IP and port (you understand that it is sensitive option when trunk is blocked by country border GW)

continue building up main server with codec conversion (will install g729/g723 codecs) amd Install OpenVPN Server&client - at this stage we will test it and measure the BW compression with all kinds of options like codecs and openvpn compression modes;

continue project with compiling the automated installation distribution (with OpenVPN, Asterisk, Codec conversion, IAX trunks config ) for client-side CentOS system, which can be distributed to may servers.

continue working on project by building up WEB interface for main server adding Billing, and other options from Item 2 like adding GW, adding client, adding IAX trunks

here I add some company we need similar thing

[url removed, login to view]

[url removed, login to view]

please contact with us ASAP if you can do this project

Skills required:

Asterisk PBX, Linux, VoIP

Skills: Asterisk PBX, VoIP

See more: unlimited web traffic, the active network, is in a prefix, gui test plan, codec conversion, asterisk pbx web gui, amd com, sip servers, open pbx, border gateway protocol, goal com, vpn exam, VPN client, voip vpn, voip termination, us exam, trunk sip, syncswitch, skills exam, sip trunk, sip provider, sip calls, side plan, pbx ip, openvpn linux

Project ID: #5000625

2 freelancers are bidding on average $1000 for this job


I have extensive experience with Asterisk codecs, Festival and IVR, please visit my site at http://www.sideeffectsystems.com, using Festival in together with a Java based webphone to synthesize country names and mress More

$1666 USD in 7 days
(1 Review)

Hi We having experience on this work. 300 kbps for 1E1(32 channenls). Also G723 &29 will support. You will see good quality of voice like syncswitch. skype:tunneluk

$333 USD in 3 days
(0 Reviews)