Asterisk/SBO Server [For VoIP Call Bandwidth Optimize]

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

We need some kind of bandwidth compression system ( up to 60-80% than usual SIP calls )from Server A to Server B.

Server A = Asterisk server

Server B = Asterisk Client server

Explanation of the 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 (quantum 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

[login to view URL]

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please contact with us ASAP if you can do this project

About the employer

Beceriler: Asterisk PBX, Debian, Linux, Sistem Yöneticisi, VoIP

Daha fazlasını gör: asterisk voip call center, asterisk voip call forward, click voip call asterisk solution, voip nat, test voip ports, allow voip through firewall, bandwidth optimizer for voip, multiple sip phones behind nat, voip, dial dialer asterisk vicidial freepbx pbx voip foip a2billing trixbox ivr fax sms gsm phone call, asterisk sbo server, low bandwidth voip call, voip call termination bandwidth, voip traffic bandwidth optimizer asterisk, sbo server bandwidth, bandwidth compression linux centos voip call, voip call bandwidth optimizer asterisk install configure linux, asterisk server voip call, voip bandwidth optimize, monitor voip call asterisk

İşveren Hakkında:
( 0 değerlendirme ) Comilla, Bangladesh

Proje NO: #18893178

Bu iş için 6 freelancer ortalamada $1743 teklif veriyor


I can do it, I have 9 years of Linux and VoIP experience and I am a Digium certified, I am sure I can do it thanks

$2500 USD in 15 gün içinde
(67 Değerlendirme)
$777 USD in 10 gün içinde
(37 Değerlendirme)

Hi , im a system administrator and voip providers manager, thanks to contact me about building your needs in a clean environnement, a web demo portal interacting with asterisk servers in a multi-tenant mode and a sip p Daha Fazla

$2500 USD in 30 gün içinde
(12 Değerlendirme)

Hi, I have over 15 years experience in VOIP. I have been doing termination in developing countries and using exactly the same technology as you are looking for. I assure you of a high quality professional service as I Daha Fazla

$700 USD in 10 gün içinde
(19 Değerlendirme)

Hi there, We have rich experience with asterisk and VOIP call apps. Please contact us so that we can help you as per your requirements. Regards, Ajit

$1500 USD in 10 gün içinde
(5 Değerlendirme)

hello, i know all software and technologies that you described in project. so, it's only question about time and money. be free to ask me any question. (continue working on project by building up WEB interface Daha Fazla

$2480 USD in 55 gün içinde
(15 Değerlendirme)