VOIP/Asterisk Expert for Developing compression Software (SBO/RBC/OSL/ZIP/OPTIMA)
$750-1500 USD
ปิดแล้ว
โพสต์ มากกว่า 9 ปีที่ผ่านมา
$750-1500 USD
ชำระเงินเมื่อส่งงาน
We are looking forward to develop an application that is currently being served by few companies (see the list below) as far we have studied, this software do tunnel between two asterisk servers to compress and bypass voice packets, we will provide some demonstration of the existing services to the candidates after interviewing and once we believed that you can do it.
Server A = Asterisk Server
Server B = Asterisk Client
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
D. Any other better disto suggested by the developer.
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 be selling this service to client as SaaS, so we will require automated license obtaining functions and authorization of the ISO upon creating new linux server (RackSpace Cloud) there are some companies already providing the solution with similar functions, PM for demonstration #
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.
Some companies already developed the solution and providing services, below is the list of competitors:
1. [login to view URL]
2. [login to view URL]
3. [login to view URL]
4. [login to view URL]
5. [login to view URL]
Please do contact if you are interested to work in the project, we will prefer if someone who have the good knowledge of what we are looking for and have some existing idea (let us know if you have already seen such thing or worked on, we will treat you on higher priority)
Thanks for checking, and looking forward to work with you!
and we can make u a commission based partner while we sell this to customers ,
Hello sir,
As I would like to tell you we are the top VOiP app developers on freelancer in india. Hence this is something that really interests us as we as a company always look forward to the project which challenges us and take us to new heights.
I understand what you are trying to do and we have already seen this technology on Nanu app if you know it. Now as I mentioned earlier we would love to collaborate with you on this for this project and would like to tell you we do have the required expertise for the same.
Here is a next gen home automation system for which we are building the VOiP solutions partner for them :- https://www.freelancer.com/projects/6867069.html
Secondly we as a company are working on a solutions which would rival the third party mobile API's provider example parse and quickblox we are not developing it completely done by us but we are the team working on VOiP and video part and chat and other part are done by other
To put the cost in prospective the requirements that you have a a system which is in itself complex to develop after that is needs to be moulded into a SaaS product which requires lots of effort we as a company follow ethics and be upfront and honest with the client so to give you the costing this would be the minimum it would take from out end.
Please discuss your project with us to so that you can know our expertise further and decide.
Kindly initiate the communications,
Regards
Team NZT
Hi there
I am expert and experienced Network planing, VoIP and Collaboration engineer, having years of resume on enterprise networks such as Mobile Career's platform and Telecom Companies.
I have worked on enterprise IVR and Speech systems and DEMO on current systems will be provided.
I have Designed, Implemented and Operated more than 10 Full termination platform, and have a pure knowledge on VoIP and fully familiar with the best soft-wares to do this. (Asterisk , A2billing, Yate Telephony)
As a resume on my local projects i have finished a termination platform for a mobile career and i can Demo for you, and the resumes on freelancer are :
https://www.freelancer.com/projects/Asterisk-PBX-network-administration/kamailio-fault-tolerant-voice-switch.html
https://www.freelancer.com/projects/Linux-System-Admin/Asterisk-with-Bandwidth-Optimizer-for.6886121.html
Looking for more information just checkout my profile.
Kindly looking forward for your information.
Regards
Hi,
We have a ready made product, which is running successfully in Pakistan, Bangladesh, Egypt, Iran and all other VoIP blocked countries. I can provide you a demo for 48 hours / 32 ports. You can check the ACD and ASR.
Please visit - pixeebytesaver
Thanks
Shan Jay
Skype : nishanthanj