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Asterisk/SBO Server [For VoIP Call Bandwidth Optimize] - Repost - repost

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

Beceriler: Asterisk PBX, MySQL, VoIP

Daha fazlasını görün: voip web server, unlimited web traffic, gui test plan, codec conversion, bandwidth com, asterisk pbx web gui, amd com, using asterisk, sip servers, open pbx, border gateway protocol, vpn exam, voip vpn, VoIP termination, VoIP PBX, us exam, trunk sip, syncswitch, sip trunk, sip server, sip provider, sip calls, pbx server, pbx ip, openvpn linux

İşveren Hakkında:
( 3 değerlendirme ) Ajman, United States

Proje NO: #4398546

5 freelancer bu iş için ortalamada 3040$ teklif veriyor

meral

hi. still can do it.

in 60 gün içinde3000$ USD
(89 Değerlendirme)
7.5
abusayed2004

I have ready solution to offer with puppy Linux...

in 3 gün içinde10500$ USD
(17 Değerlendirme)
5.2
jamesvos

Able to fill-up your almost requirements.

in 3 gün içinde600$ USD
(0 Değerlendirme)
0.0
Genpex0

Dear Sir, I'm a VOIP engineer and have more than 7 years of experience in implementing and managing asterisk systems. I have worked in Asterisk Distros like Elastix/TrixBox/PIAF/AsteriskNow....etc. Also Implemen Daha fazlası

in 5 gün içinde550$ USD
(0 Değerlendirme)
0.0
attocom

Atto communication pvt ltd based out in Bangalore, India. We started in 2011 with team of people who have rich experience Marketing , Software development and Testing in field of Networking and Wireless Daha fazlası

in 3 gün içinde550$ USD
(0 Değerlendirme)
0.0