SIP balancer

From Kolmisoft Wiki
Revision as of 06:47, 5 February 2014 by Nerijuss (talk | contribs)
Jump to navigationJump to search

SIP balancer is available starting from MOR X4 and is available only with multiple Asterisk server solutions. Technology of SIP balancer is based on the OpenSIPS


Benefits

  • Provide one unique hostname or IP for your providers/clients
  • Handle multiple DID numbers from different providers by sending them to one IP address
  • Distribute calls evenly to servers depending on the load each server can handle
  • Scalable architecture - you can add new Asterisk servers at any time or remove them

Features

  • Handle incoming calls from DIDs, trunks, providers
  • Monitoring of Asterisk server capacity - allows to set how many calls a certain Asterisk server can handle
  • If one Asterisk server becomes unresponsive or reached call limit, no more calls are sent to that server
  • A simple GUI to manage the SIP balancer.

Architecture

Possible implementation scenarios can be found here

Logic

  • All registrations and register requests are sent to the SIP balancer directly
  • In order to call any device it must be called through a DID (DID can be fake inside the system), otherwise calls

between devices will not work. When a call is made through a DID to device, that call is sent to proxy as it handles the SIP registrations. Same applies to trunks, so trunks should authorize calls from the proxy IP.

  • Outgoing calls to providers are sent from Asterisk servers, so your provider should allow

calls from the Asterisk server IPs.

  • Incoming calls from providers should be sent to Proxy, but the system could be adjusted to

send accept them on Asterisk servers too.

  • All media requests are handled by Asterisk servers

Recommended hardware

It should be similar to Recommended hardware for MOR server with minimum 2 GB RAM.

Such solution would be suitable to load the balance between 2-3 asterisk servers.

For more servers you should use more powerful hardware.

Configuration

Accept calls from any port

When SIP balancer is in use and Device/Provider type is SIP and it is IP Authenticated then near to Port input you can find "Accept calls from any port" setting.

1. By default pressed No and you can type any port you want

2. When Yes is pressed, you are not able to type any port and calls will be accepted from any port.

Allow calls from any port.png

See also

Retrieved from "https://wiki.kolmisoft.com/index.php?title=SIP_balancer&oldid=17371"