>>> using Asterisk 1.4.28, I encountered a problem with SIP >>> RTP port allocation. >>> >>> I found some entries in mailinglist and bugtracker regarding >>> this issue, but only old ones. >>> >>> My rtp.conf has >>> [general] >>> rtpstart=30000 >>> rtpend=30100 >>> >>> so 100 ports available. I know that up to 4 ports per channel can be used >>> and so up to 25 channels are possible. > 4 ports only if you use audio and video. We use two ports per RTP stream - > and send on two ports, but this is for incoming media. > So 100 ports is enough for 50 audio calls.
Even if it isn't a video call, I think as soon as videosupport is activated, the additional 2 ports are allocated. >>> But even earlier I often get the error about "No RTP ports remaining". >>> >>> I had a look at >>> netstat -nuap >>> and it shows that a lot of ports are still assigned, even if there is no >>> channel in use. >>> But "sip show channels" show a lot of (unused) entries with no >>> codec/Format and "Last Message" like INVITE, REGISTER, OPTIONS. > REGISTER and OPTIONS allocate no RTP ports, so those are not a problem. If > you have a SIP channel that has a last message being INVITE and still say > you have no calls, you have a problem right there. I just see these entries with "sip show channels", but cannot tell if e.g. the REGISTER listed channels have RTP ports allocated. Who can I find out which SIP channel allocated which port? Or which SIP channel belongs to the ports I see with 'netstat -nuap'? I do have a sip channels like 172.21.4.114 666 0430c3a638e 00102/00000 0x0 (nothing) No Init: INVITE in 'sip show channels' and they don't go away for a long time. Shouldn't there be a timeout to destroy such a channel even if somehow the phone was 'disconnected' in during a call? >> If the channels exists even after 32 seconds after BYE, and BYE was >> signaled correctly, I would file a bug report. > > Yes, the RTP ports should be closed at least at that point, when we destroy > the SIP channel. Anything else is a bug. I am not really sure about when > they're closed, but I'm trying to understand that in my RTCP adventures > since I want to change it. Before filing a bug, I would like to be sure that I have checked all possibilities here. To me it looks like that some special event leaves a sip channel activ and not be destroyed. So when Asterisk runs for a longer time, more and more channels like this occur. Ayn idea how to check this? Armin -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users