SIP balancer
From Kolmisoft Wiki
Jump to navigationJump to search
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
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
- 4 server redundant solution
- 2 server redundant solution
- Implementations
- For pricing quotes contact our sales department