SIP balancer

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.

See also

• Implementations
• For pricing quotes contact our sales department