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Low bandwidth voip solution by asterisk.

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 .

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 Noman

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.

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.

for your understand . here i give you a web site . as they provide .. we need same solution,
http://www.syncswitch.com/

check the site . we need same solution.

waiting for your update.

regards,
Md. Abu Noman



Kỹ năng: Asterisk PBX, Linux, VoIP, Windows Server

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( 0 nhận xét ) Kulalaumpr, Bangladesh

Mã Dự Án: #1706124

6 freelancer đang chào giá trung bình $2333 cho công việc này

meral

hi. can do central server and rpm for outer servers(centos 6 setup required). bid not include website work, dialler [url removed, login to view] web for control remove servers and setup saving trunk. i work in voip fro 9 years and i am b Thêm

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shakoush2001

Hi I am a CISSP,RHCE,CCNA,MCSA,Linux+ and a CEH. I do have 7+ years experience in System Administration . I have experience in a high availability environment with 100+ servers and more than 500 000+ subscribers, I kno Thêm

$1500 USD trong 20 ngày
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ekoandriprasetyo

Ready to work.

$750 USD trong 3 ngày
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K5Sg8h7OE

Custom software development - <b><i>Removed by Admin</i></b>

$750 USD trong 1 ngày
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masterasterisk

Dear Hiring Manager, I have an Expertise in Voice Broadcast, VPS, Trixbox, FlashPBX, FreePBX, GoAutoDial, Asterisk, Elastix, IVR, A2billing with experience in migrating applications to the cloud, and I'm very intere Thêm

$1500 USD trong 21 ngày
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mohitshrm334

Hi, I have more than 7 tears of experience of doing the similar tasks, I can do this asap. Regards Anil

$4500 USD trong 15 ngày
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1.8