<div dir="ltr">Thanks Yossi,<div>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.</div>
<div><br></div><div style>Am I missing something? Is there a way to dynamically create proxies for routing each call in OpenSIPS?</div></div><div class="gmail_extra"><br clear="all"><div><div dir="ltr">Thank you<div><i><font color="#0000ff">Oleg</font></i></div>
</div></div>
<br><br><div class="gmail_quote">On Mon, May 13, 2013 at 4:33 PM, I put the Who? in Mishehu <span dir="ltr"><<a href="mailto:mishehu@freeswitch.org" target="_blank">mishehu@freeswitch.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
<div>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.<br>
<br>
<a href="http://www.opensips.org" target="_blank">http://www.opensips.org</a><br>
<a href="http://www.kamailio.org" target="_blank">http://www.kamailio.org</a><br>
<br>
-Yossi<div><div class="h5"><br>
<br>
On 05/13/2013 10:40 AM, Oleg Stolyar wrote:<br>
</div></div></div>
<blockquote type="cite"><div><div class="h5">
<div dir="ltr"><span style="font-family:arial,sans-serif;font-size:13px">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)</span>
<div style="font-family:arial,sans-serif;font-size:13px">
<br>
</div>
<div style="font-family:arial,sans-serif;font-size:13px">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?</div>
<div style="font-family:arial,sans-serif;font-size:13px"><br>
</div>
<div>
<div dir="ltr">Thank you
<div><i><font color="#0000ff">Oleg</font></i></div>
</div>
</div>
</div>
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