Thanks Yossi,
I looked into OpenSIPS - it's an option but it seems to also load the media server instances on load, which is something I think I can do with FreeSWITCH as well, so no reason to use two solutions.  FreeSWITCH has its own SIP (Sofia) component which I was hoping to use to route SIP calls to other instances of FreeSWITCH.

Am I missing something?  Is there a way to dynamically create proxies for routing each call in OpenSIPS?

Thank you
Oleg


On Mon, May 13, 2013 at 4:33 PM, I put the Who? in Mishehu <mishehu@freeswitch.org> wrote:
SIP is the signalling, and RTP is the media stream.  For load-balancing of SIP calls to different servers, you're probably better served by OpenSIPS or a similar project (i.e. kamailio).  FreeSWITCH can readily interoperate with OpenSIPS and Kamailio.

http://www.opensips.org
http://www.kamailio.org

-Yossi


On 05/13/2013 10:40 AM, Oleg Stolyar wrote:
I am very new to FreeSWITCH and would like to use it as a SIP load balancer to other (media) FreeSWITCH instances.  I was thinking of using the external SIP profile and bridging incoming calls to the media instances of FreeSWITCH using some rotation mechanism (round robin, least busy, etc)

My question is this:  Would I need to have a separate SIP profile for each media FreeSWITCH server or can I define a profile in such a way that I can dynamically specify the proxy for each call?

Thank you
Oleg


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_________________________________________________________________________
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http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
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http://wiki.freeswitch.org
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