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loadblancing & failover configuration for asterisk boxes

we want some experienced engineer to work out the following loadbalancing & failover configuration for us,

1.2 to 5 asterisk boxes

[url removed, login to view] or kamilio frontend to interface with one or more sip trunks for incoming calls, and then routing the calls incoming to the asterisk boxes with load balancing mode, and bypass the failed asterisk boxes

[url removed, login to view] phones are sip phones basically, they can register to asterisk boxes via the opensips or kamillio box , so the registeration can be evenly distributed to the asterisk boxes, and bypass the failed asterisk boxes

[url removed, login to view] opensips / kamilio server should be able to fail over to another server when fault happen, and kept on working , the system interruption should be less than 10 seconds

[url removed, login to view] opensips / kamillio server may need to do protocol conversion for calls from sip trunks , from tcp to udp or vice versa

[url removed, login to view] opensips / kamillio should count the transactions or dialogs of the sip registration and call processes, not just count the sip messages, so it would know when the registeration / call start & end , and can have the correct count of the load on each asterisk server.

[url removed, login to view] opensips / kamillio should check the live status of asterisk by polling with sip options message or other workable ways

[url removed, login to view] the opensips / kamillio server could send the current load change information to another socket based program , it would be a plus.

[url removed, login to view] opensips / kamillio server should be able to h andle 1000 calls within 60 seconds, our servers are all Dell R310, and with 4GB RAM & one XEON X3430 CPU configured. it seems not a problem from online performance references, but you would prepare the testing scripts & guides based on sipp to simulate the incomming calls & the agent phone registering, and eventually verify the performance.

The asterisk server would be 1.4.x. , and we will setup it for you. The vpn or ssh access would be setup for the selected engineer

when you bid, pls respond all my above requests one by one, in case of misunderstanding this task.

Kĩ năng: Asterisk PBX, Linux, Thiết lập Bản thảo, Quản trị hệ thống, VoIP

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( 0 nhận xét ) Guangzhou, China

ID dự án: #1016025