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Asterisk/SBO 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 ( 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

Kemahiran: Asterisk PBX, Linux, VoIP

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Tentang Majikan:
( 0 ulasan ) Egypt

ID Projek: #4066662

8 pekerja bebas membida secara purata $1338 untuk pekerjaan ini

captek

Thanks for explanation. Hi, We are asterisk/freeswitch/VoIP solution provider and have setup various server for calling, load balancing, redundancy, media relay with features like sip call, video call, fax, ivr etc. We Lagi

$1500 USD dalam 10 hari
(10 Ulasan)
6.5
abusayed2004

I can do it, I have similar experience and running server with bandwidth optimization technology.

$1500 USD dalam 7 hari
(14 Ulasan)
4.4
saifcce

Dear, I am working in a IP Telephony Provider as an Asst System Admin.I successfully complete almost twenty Asterisk project in Odesk. Please See my profile for details. https://www.odesk.com/users/~01e3b2f97f87b40 Lagi

$200 USD dalam 3 hari
(10 Ulasan)
3.7
scriptlancerinbd

Hi! Probably you are looking for something I have! --SG

$2000 USD dalam 15 hari
(1 Ulasan)
1.4
moustafaham

Greetings, Kindly check PMB

$1500 USD dalam 5 hari
(0 Ulasan)
0.0
vw7237226vw

Hello, I can handle this project for you. I have extensive experience with Asterisk as well as OpenVPN. I have run VOIP calls over low-bandwidth/high latency VSAT links. I have also developed remote provisioning Lagi

$1500 USD dalam 25 hari
(0 Ulasan)
3.6
axenicglobal

Please check the personal message for company detail

$1500 USD dalam 30 hari
(0 Ulasan)
0.0
razib

HI, I have good experience in asterisk, I can to do this

$1000 USD dalam 15 hari
(0 Ulasan)
0.0