Hurray for Microsoft Outlook (for creating this whole top-post thread). Just my .02; The other two ports must have been a remnant of another channel; as for the 4 ports - I think that the 4 port requirement is probably for "niceties" like conferencing and transfers.
_____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B Sent: Friday, January 14, 2011 2:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Why are 4 ports used for a single call? 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 <solsta...@gmail.com> 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 <bruceb...@gmail.com> wrote: I mean part of RTP RFC? On Fri, Jan 14, 2011 at 2:41 PM, Bruce B <bruceb...@gmail.com> 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
-- _____________________________________________________________________ -- 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