Vovida.org load balancer

This project has been taken over by CMSOFAZ.COM.

I have updated the project to work with the newest version of asterisk (1.2.7.1), and added the following features.
After updating the project I ran it in production for 3 days to verify that it worked without any issues.

- Fixed the way it talked SIP
- Added Live Registry File to see where a user is
- Added Extension for libProxy to login as
- removed windows support

Send comments / Suggestions to info@cmsofaz.com.

DOWNLOAD


Quickly Get Started:

Compile source

Put this in your asterisk sip.conf
[9999]
type=friend
host=dynamic
username=9999
qualify=yes
context=default


and run with
Sample usage: /sbin/lbProxy -name 172.16.51.9 -reqPort 5060 -respPort 5061 -proxy 172.16.50.100:5060 -proxy 172.16.50.1:5060 -proxy 172.16.51.34:5060 -extension 9999 -registryFile /var/log/registry


From the README:



The Load Balancer is a very simple proxy that is useful in SIP-based VoIP installations where there are multiple ingress proxy servers.
The Load Balancer permits pooling these servers, thereby eliminating the need to balance user demands for connectivity through a complicated provisioning algorithm. All users can send their INVITEs and REGISTERs to the same SIP URI and the Load Balancer will assign ingress proxy servers dynamically to each transaction. In this way, the traffic load is balanced over a pool of proxy servers based on the real-time demand for services.

The Load Balancer receives SIP requests from endpoints on a port that is specified in the configuration file(?). It then forwards the request to the next available ingress proxy server that appears on a list of associated proxy servers that is created at compile time(?). The Load Balancer receives responses on a different port, and then forwards them back to the originators of the requests.

The Load Balancer adds itself to the Via header of requests to enable responses to return before being sent to orginating endpoint. This only works with SIP messages sent over UDP (User Datagram Protocol).




This project has been taken over by CMSOFAZ.COM.

I have updated the project to work with the newest version of asterisk (1.2.7.1), and added the following features.
After updating the project I ran it in production for 3 days to verify that it worked without any issues.

- Fixed the way it talked SIP
- Added Live Registry File to see where a user is
- Added Extension for libProxy to login as
- removed windows support

Send comments / Suggestions to info@cmsofaz.com.

DOWNLOAD


Quickly Get Started:

Compile source

Put this in your asterisk sip.conf
[9999]
type=friend
host=dynamic
username=9999
qualify=yes
context=default


and run with
Sample usage: /sbin/lbProxy -name 172.16.51.9 -reqPort 5060 -respPort 5061 -proxy 172.16.50.100:5060 -proxy 172.16.50.1:5060 -proxy 172.16.51.34:5060 -extension 9999 -registryFile /var/log/registry


From the README:



The Load Balancer is a very simple proxy that is useful in SIP-based VoIP installations where there are multiple ingress proxy servers.
The Load Balancer permits pooling these servers, thereby eliminating the need to balance user demands for connectivity through a complicated provisioning algorithm. All users can send their INVITEs and REGISTERs to the same SIP URI and the Load Balancer will assign ingress proxy servers dynamically to each transaction. In this way, the traffic load is balanced over a pool of proxy servers based on the real-time demand for services.

The Load Balancer receives SIP requests from endpoints on a port that is specified in the configuration file(?). It then forwards the request to the next available ingress proxy server that appears on a list of associated proxy servers that is created at compile time(?). The Load Balancer receives responses on a different port, and then forwards them back to the originators of the requests.

The Load Balancer adds itself to the Via header of requests to enable responses to return before being sent to orginating endpoint. This only works with SIP messages sent over UDP (User Datagram Protocol).




Created by: oej, Last modification: Thu 14 of Jun, 2012 (19:44 UTC) by admin
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