Technology for reducing bandwidth consumption of VoIP to a great extent

• Technology for reducing bandwidth consumption of VoIP to a great extent (up to 84%).

• Uses lossless compression method, thus can be used with mostly used codecs i.e. [url removed, login to view], [url removed, login to view], GSM etc.

• Works from NAT - from real IP and/or public IP, static and/or DHCP.

• No change in voice quality and route statistics, rather it can increase Average Call Duration (ACD) to 1 minute.

• Works as a third party device/software with any Soft Switch.

• “Point-To-Multi Point” technology, requires native devices at both end. It works as like tunnel to secure packets and bypass all kind of firewall.

• SUPER BANDWIDTH OPTIMIZER uses lossless compression method and do not use codec for providing better compression, rather it compresses transport (RTP). So it works with all commonly used codecs i.e. [url removed, login to view], [url removed, login to view], GSM, AMR etc.

• It works with Session Initiation Protocol (SIP

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 can provide you the Dedicated server asterisk and client asterisk as for model.

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

Evner: Asterisk PBX, Linux, VoIP

Se mere: voip web server, unlimited web traffic, is in a prefix, gui test plan, codec conversion, building super, voip server software, voip firewall, sip servers, server technology, conversion optimizer, border gateway protocol, vpn exam, voip vpn, VoIP termination, voip switch, voip rtp, trunk sip, soft switch, sip trunk, sip provider, sip calls, secure voice, rtp, openvpn linux

Om arbejdsgiveren:
( 0 bedømmelser ) France

Projekt ID: #6047163

12 freelancere byder i gennemsnit $2891 på dette job


hi there are no way do lossless compression. gsm,g723,g729 are ALL lossless!!. also 80% can be archived only if you compress ulaw to g729. all other variants will be much less compression. 84% is not achivable withou Flere

$2684 USD in 30 dage
(175 bedømmelser)

Hi, I have a really good experience in VoIP networking. I have develop a solution with those specifications. Please take a look at: [login to view URL] Flere

$2500 USD in 30 dage
(5 bedømmelser)

Hello, I can help you to do this I am an expert on voip asterisk and vpn . if you are looking for a professional engineer. stop looking because you found the right person for the right place. Contact me. I Flere

$2222 USD in 30 dage
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I have extensive experience that meets the qualifications, and would be happy to complete this work for you. We can speak via telephone ( 2569192001), skype - cesurasean, yahoo im, aim, msn messenger, or irc chat. T Flere

$2500 USD in 30 dage
(2 bedømmelser)

Hello. I can work for you. I've experienced in C, C++, Network, OS, Asterisk development. As you know, I think the iax trunking is most to reducing bandwidth. And about codec, did you consider silk codec (support from Flere

$1500 USD in 20 dage
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Hi, thanks for your job posting. I’v good skill on iOS and android app developing with 6 years XP. Please check my portfolio and touch base with me. I am ready to start and waiting for your response. Thanks. Regar Flere

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Hi, Webmyne is ISO 9001:2008 certified India & Canada based since 2002. (New On I have reviewed your requirement and sure that we can provide you best solution because We had work and successfu Flere

$2500 USD in 30 dage
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$5263 USD in 30 dage
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Hello, Sir Thanks for your great job. I am very profession at voip projects. I have been developing multimedia communication projects for 6 years. Please check my voip app. [login to view URL] Flere

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Hi this is Deval from KCSPL IT Software Consulting Company". As glad to share further information about our Technological Capabilities, kindly have a look on our Website: [login to view URL] We have developed Flere

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Hello, We are the leading Software consulting company based at India. We work on high end technologies and having sound exposure of working on the complex projects of Web, Mobile, VOIP, Conference, Asterisk, IVR, PB Flere

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Hi there , We have gone through your initial project description and based on your requirement we want to go for detail analysis of your requirement and brief discussion with [login to view URL] find below our Basic skill in IT Flere

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