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Asterisk/SBO Server [For VoIP Call Bandwidth Optimize]

$30-250 USD

Closed
Posted about 9 years ago

$30-250 USD

Paid on delivery
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 [login to view URL] [login to view URL] please contact with us ASAP if you can do this project
Project ID: 7098063

About the project

10 proposals
Remote project
Active 9 yrs ago

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10 freelancers are bidding on average $4,135 USD for this job
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Our team is specialized in VoIP development, VoIP and Asterisk especially is our main specialization. We have wide experience of building customized solutions, fully based on open source products. My bid is approximate, because we have to discuss your list of requirements first. We are located in UTC+2 time zone.
$5,555 USD in 3 days
5.0 (4 reviews)
5.3
5.3
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A proposal has not yet been provided
$15,555 USD in 3 days
4.9 (25 reviews)
4.6
4.6
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A proposal has not yet been provided
$277 USD in 15 days
4.9 (17 reviews)
4.3
4.3
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A proposal has not yet been provided
$750 USD in 15 days
5.0 (4 reviews)
3.4
3.4
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Seems challenging
$833 USD in 15 days
5.0 (4 reviews)
2.8
2.8
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Hi, I work with severall VOIP carriers, and the setup you need we use it a lot. there are som cons in iax trunking, if the connection looses packets, it will drop all the calls. anyway it is the best solution for saving bandwidth. Also I would recommend yate for server B and I will have to consider the hardware size according to the number of calls you have on the other end. We had a core2 of 3Ghz with 3e1 I actually have a wholesale gui for asterisk made from scratch, it works with nagios for monitoring and alerting. I do have several plugins for quintums and hypermedia, to individually check the health of the channels. regards
$2,222 USD in 3 days
5.0 (1 review)
2.0
2.0
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Hola soy un ingeniero de sistema dedicado a la prestación de servicio de voip en los últimos 13 años e estado dedicado al diseño y configuración de sistemas de Voip bajo plataforma Asterisk, Quintun analogos o digitales, softswichet Mera MVTS y Nex-Tone siendo proveedor al mayor y detal de terminación de llamadas LDI y LDN. En consecuencia me agradaría mucho ayudarle en este proyecto que es muy interesante Hable perfecto Español
$8,888 USD in 15 days
0.0 (0 reviews)
0.0
0.0

About the client

Flag of BOLIVIA
La Paz, Bolivia
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0
Member since Jun 6, 2014

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