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USB Bandwidth Optimizer for VOIP calls - Repost - open to bidding

$250-750 USD

In Progress
Posted over 9 years ago

$250-750 USD

Paid on delivery
Project Description: 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) 6. For server B need portion of asterisk web gui for adding gateways..prefix or so, 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 2. we need some kind of bandwidth compression system( upto 60-80% than usual SIP calls in g723.1 codec), from Server A to Server B. 3. server A to Server B voice traffic will be encrypted so that voice port blocked bandwidth can be used for termination Pls let us know for further information necessery to get the project done. This project actually similar to a provider [login to view URL] offering and we we want to provide the same kinda services to our termination provider or tht meant named Server B in our given specifications above. suggest us required Server configurations n OS for Server A to start the projects
Project ID: 6362848

About the project

4 proposals
Remote project
Active 10 yrs ago

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I have done several projects like this. Your described code g723.1 is not supported by server VoIP Dialer company, be sure before process this project. Although this project will support wide range of codec such as ulaw, alaw, gsm, g729,speex, siren14, g722, siren7, ipc10, g723, slin, g726, ilbc and many more. Server configuration depend on call processing load. Normally for 200 concurrent call it is recommended to a octa core processor powered dedicated server with 16 GB RAM. We can also help you to taking a server or we can offer you proper server depend on load. If you want I can show you a running server with this VPN or bandwidth optimization server. Feel free to contact me on skype [login to view URL] or yahoo [login to view URL]
$290 USD in 1 day
0.0 (0 reviews)
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4 freelancers are bidding on average $761 USD for this job
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Hi, We have 8 year + experience with Open source CTI tools, like Asterisk, SER, Kamailio, Freeswitch etc.. And we are fully confident about your satisfaction if you choose us for this projects. We will prepare both A,B (bootable from usb) servers by using following technallogies. 1. Linux ( CentOs ) 2. Live CD 3. RPM / Yum 4. Asterisk 5. PHP / MySQL ( for GUI ) Note : buyer will be responsible for codec licensing Please contact for further details. Regards
$1,000 USD in 10 days
4.5 (15 reviews)
5.9
5.9
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Hello, we are professionally connected with VoIP setup & consultancy. Please visit www.tunnelsolutions.co. Thanks
$555 USD in 10 days
0.0 (0 reviews)
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I have great skills in the areas of Asterisk PBX and Software Architecture due to my past experience working as a Asterisk PBX Developer and Software Architecture Developer. I am looking to be outsourced by you for your jobs so I can bring my skills and experience to you. I am also highly proficient in VoIP. If given a chance to be part of your team I will work best and bring in the required output. & we have build already this type of solution its for bandwidth optimization solution we will provide you online billing for admin & customer, so after you purchase from us you need nothing anymore just install on server & use, if you want to know more details we can discuse on S | K | Y | P | E my I D is asterdeve
$1,200 USD in 3 days
0.0 (0 reviews)
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About the client

Flag of BAHRAIN
Manama, Bahrain
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Member since Aug 21, 2014

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