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Bandwidth optimization voip solution

we need basically bandwith optimization. in client end they have termination gateways, each gateway can carry 30 simultaneous calls. and all gateway under DHCP network . we take a server which run in static IP . all client should send calls to server IP from his SIP server and we need to create separate account for each gateway . and calls should send to specific termination .

call should pass with sip / iax2 , g729 or g723. in client end we want to setup asterisk module which can convert calls from sip / IAx to sip and pass the calls to gateway .

01. asterisk or SBO server. which receive calls from many sip server.

02. asterisk./SBO transfer calls to local PC or router .

03. in router / pc have a module which can route calls to LAN IP .

04. in LAN IP there have a termination gateway . so calls can pass normal .

the thing is , u need to work on asterisk/ SBO server and client end PC/ router.

for your understand . here i give you a web site . as they provide .. we need same solution,

[url removed, login to view]

check the site . we need same solution.

waiting for your update.

regards,

Md. Abu NomanAdditional Project Description:

06/20/2012 at 2:51 EDT

1 # VOIP Call Termination Over Local IP ( VOIP Over NAT Traversal Or Firewall )

2 # VOIP Call & Quality Voice Termination In Lower Bandwidth ( 1 Call 6-8kb )

3 # Web Based Administrator, Agent & Client Control ( Easy User Interface)

4 # Easy New Gateway Add System For Client ( Add Gateway:- PC MAC Address & Gateway Local IP )

5 # Stay Alive Internet System ( As Like Skype )

6 # Easy Asterisk Billing System

7 # Easy Bootable USB Client

8 # VOIP Server Connect The Common Configuration Client Over Local IP & MAC Address.

06/20/2012 at 2:53 EDT

well ,

we need basically bandwith optimization. in client end they have termination gateways, each gateway can carry 30 simultaneous calls. and all gateway under DHCP network . we take a server which run in static IP . all client should send calls to server IP from his SIP server and we need to create separate account for each gateway . and calls should send to specific termination .

call should pass with sip / iax2 , g729 or g723. in client end we want to setup asterisk module which can convert calls from sip / IAx to sip and pass the calls to gateway .

01. asterisk or SBO server. which receive calls from many sip server.

02. asterisk./SBO transfer calls to local PC or router .

03. in router / pc have a module which can route calls to LAN IP .

04. in LAN IP there have a termination gateway . so calls can pass normal

05. VOIP Call Termination Over Local IP ( VOIP Over NAT Traversal Or Firewall )

06. VOIP Call & Quality Voice Termination In Lower Bandwidth ( 1 Call 6-8kb )

07. Web Based Administrator, Agent & Client Control ( Easy User Interface)

08. Easy New Gateway Add System For Client ( Add Gateway:- PC MAC Address & Gateway Local IP )

09. Stay Alive Internet System ( As Like Skype )

10. Easy Asterisk Billing System

11. Easy Bootable USB Client

12. VOIP Server Connect The Common Configuration Client Over Local IP & MAC Address. .

the thing is , u need to work on asterisk/ SBO server and client end PC/ router.

Skills: VoIP

See more: voip web server, voice over description, md web, edt convert, bandwidth com, agent module, voip firewall, skype bandwidth, VoIP termination, voip server, voip bandwidth optimization, syncswitch, sip router, setup voip solution, bootable, bandwidth, bandwidth optimization, asterisk billing, sip administrator, bandwith sbo, server configuration voip, iax2 sip, asterisk bandwidth work, voip web sip, g729 sip client

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Project ID: #5453158

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