Difference between revisions of "SIP balancer"

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[[Image:SIP_balancer_for_docs.png]]
[[Image:SIP_balancer_for_docs.png]]
== 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 ==
== Recommended hardware ==

Revision as of 09:24, 7 March 2013

Currently KOLMISOFT offers a SIP balancer with multiple Asterisk server solutions.
The 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.

Current Limitations (will be resolved upon request):

Architecture

SIP balancer for docs.png

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 to asterisk servers.

For more servers you should use more powerful hardware.

See also