Each media stream will use two, one for RTP and one for RTCP. In your case 10200/10201 are an RTP/RTCP pair, and 10504/10505 are another pair. RTP is always and even numbered port, and RTCP is always RTP port + 1. Yes, it's in the RFC for RTP.
The fact that you have two pairs means that two media streams are being negotiated, perhaps one for audio and one for video? Your phone config or wireshark captures will tell you for certain. Of course, I'm assuming that those ports are for one endpoint (phone). If one pair is for caller and one pair is for callee, then this is a normal "simplest" scenario, one pair for each side. You didn't specify "whose" ports they were. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B Sent: Friday, January 14, 2011 11:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Why are 4 ports used for a single call? 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
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