<html>
<head>
<meta content="text/html; charset=utf-8" http-equiv="Content-Type">
</head>
<body bgcolor="#FFFFFF" text="#000000">
<p>No there is no such thing as magic.</p>
<p>The most obvious way to implement the RTP port handling, is to
first open the next UDP port in the OS, and then report that back
in the Invite/200Ok. If the port cannot be opened, then simply try
the next in line. <br>
</p>
<p><br>
</p>
<pre class="moz-signature" cols="72">Med venlig hilsen / Best regards
Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
Viptel ApS, Hammershusvej 16C, DK-7400 Herning
Telefon: +45 46949949, Telefax: +45 46949950, <a class="moz-txt-link-freetext" href="http://viptel.dk">http://viptel.dk</a></pre>
<div class="moz-cite-prefix">On 03/13/2017 01:52 PM, przeqpiciel
wrote:<br>
</div>
<blockquote
cite="mid:CADgxv4rzuf5orOCaOv0HuA0hUebeYTq9Hm-kUs0=qLvLYWXi5Q@mail.gmail.com"
type="cite">
<meta http-equiv="Content-Type" content="text/html; charset=utf-8">
<div dir="auto">Maybe there is an magic device? I know that if we
have an asterisk, that become to us with default configuration
of rtp ports sets to 10000_20000. And each call choose the one
port fron that range. So if we have several asterisks with
default configuratiin of rtp, there is possibilities to have 2
concurent calls each through another asterisk instance with this
same rtp port. Am i right?
<div dir="auto"><br>
</div>
<div dir="auto">So mqybe this magic device could see source IP
address and route rtp to correct adterisk?</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">13.03.2017 7:15 AM "Alex Balashov" <<a
moz-do-not-send="true"
href="mailto:abalashov@evaristesys.com">abalashov@evaristesys.com</a>>
napisaĆ(a):<br type="attribution">
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">On Mon,
Mar 13, 2017 at 07:08:09AM +0100, Kjeld Flarup wrote:<br>
<br>
> We run multiple Asterisk instances since 1.4 and never
configured RTP ports.<br>
><br>
> More challenging issues are the Asterisk DB, and the
Asteisk home.<br>
<br>
You may not have enough calls for RTP port collisions to
become an<br>
issue. Otherwise, I'm not sure how you're avoiding it, since
Asterisk<br>
isn't aware of which ports from within the range are in use.<br>
<br>
--<br>
Alex Balashov | Principal | Evariste Systems LLC<br>
<br>
Tel: <a moz-do-not-send="true" href="tel:%2B1-706-510-6800"
value="+17065106800">+1-706-510-6800</a> / <a
moz-do-not-send="true" href="tel:%2B1-800-250-5920"
value="+18002505920">+1-800-250-5920</a> (toll-free)<br>
Web: <a moz-do-not-send="true"
href="http://www.evaristesys.com/" rel="noreferrer"
target="_blank">http://www.evaristesys.com/</a>, <a
moz-do-not-send="true" href="http://www.csrpswitch.com/"
rel="noreferrer" target="_blank">http://www.csrpswitch.com/</a><br>
<br>
______________________________<wbr>_________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing list<br>
<a moz-do-not-send="true"
href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>
<a moz-do-not-send="true"
href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users"
rel="noreferrer" target="_blank">http://lists.sip-router.org/<wbr>cgi-bin/mailman/listinfo/sr-<wbr>users</a><br>
</blockquote>
</div>
</div>
<br>
<fieldset class="mimeAttachmentHeader"></fieldset>
<br>
<pre wrap="">_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a>
<a class="moz-txt-link-freetext" href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a>
</pre>
</blockquote>
<br>
</body>
</html>