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<div class="moz-cite-prefix">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 class="moz-txt-link-freetext" href="http://www.opensips.org">http://www.opensips.org</a><br>
<a class="moz-txt-link-freetext" href="http://www.kamailio.org">http://www.kamailio.org</a><br>
<br>
-Yossi<br>
<br>
On 05/13/2013 10:40 AM, Oleg Stolyar wrote:<br>
</div>
<blockquote
cite="mid:CA+x4QzA_Mn_52nKWuauBVKYO_4uucvCftgVaOxBM+aOyFpzM1Q@mail.gmail.com"
type="cite">
<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>
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