Thanks guys. I am not sure whether that call was asymmetric or not but I saw 4 ports open. It could be that the other two ports were <span class="Apple-style-span" style="font-family: arial, sans-serif; ">remnant</span><span class="Apple-style-span" style="font-family: arial, sans-serif; "> </span>of another channel even though I doubt it. <div>
<br></div><div>Now, when I tried again, it is only 2 ports that is opened like you mentioned, even RTP port, and RTP port +1. So, does Asterisk usually use the symmetric method or is the asymmetric method used as well by some media servers? </div>
<div><br></div><div>The reason why I am asking is because there are many many online responses that there is 4 ports needed per call and make sure you keep enough ports open, blah blah...</div><div><br></div><div>Thanks again<br>
<br><div class="gmail_quote">On Fri, Jan 14, 2011 at 2:57 PM, Gary Allen <span dir="ltr"><<a href="mailto:solstars1@gmail.com">solstars1@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
RTP always uses a random even numbered port, then RTCP will use the next port, which will always be odd numbered. Symmetric RTP only needs two ports, while asymmetric RTP uses four.<br><br><a href="http://www.armware.dk/RFC/rfc/rfc4961.html" target="_blank">http://www.armware.dk/RFC/rfc/rfc4961.html</a><br>
<br><br><br><div class="gmail_quote"><div><div></div><div class="h5">On Fri, Jan 14, 2011 at 12:44 PM, Bruce B <span dir="ltr"><<a href="mailto:bruceb444@gmail.com" target="_blank">bruceb444@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);padding-left:1ex"><div><div></div><div class="h5">
I mean part of RTP RFC?<div><div></div><div><br><br><div class="gmail_quote">On Fri, Jan 14, 2011 at 2:41 PM, Bruce B <span dir="ltr"><<a href="mailto:bruceb444@gmail.com" target="_blank">bruceb444@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);padding-left:1ex">
Hi Everyone,<div><br></div><div>I am just tweaking a pfSense router and learning lots about NAT etc....I noticed that each call uses four UDP port for RTP. Here is an example of port for a call I made:</div><div><br></div>
<div><div>10200</div><div>10201</div><div>10504</div><div>10505</div></div><div><br></div><div>Seems like they are random in pair. I have a restriction of 10000-11000 in my rtp.conf so that makes sense. But why use 4 ports per call? is that part of SIP RFC?</div>
<div><br></div><div>Thanks</div>
</blockquote></div><br>
</div></div><br></div></div>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br><div>
</div>
<br>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br></div>