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 remnant of another channel even though I doubt it.
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? 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... Thanks again On Fri, Jan 14, 2011 at 2:57 PM, Gary Allen <[email protected]> wrote: > 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. > > http://www.armware.dk/RFC/rfc/rfc4961.html > > > > On Fri, Jan 14, 2011 at 12:44 PM, Bruce B <[email protected]> wrote: > >> I mean part of RTP RFC? >> >> >> On Fri, Jan 14, 2011 at 2:41 PM, Bruce B <[email protected]> wrote: >> >>> Hi Everyone, >>> >>> 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: >>> >>> 10200 >>> 10201 >>> 10504 >>> 10505 >>> >>> 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? >>> >>> Thanks >>> >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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