Am 13.02.2010 09:26, schrieb Olle E. Johansson: > > 12 feb 2010 kl. 16.43 skrev Klaus Darilion: > >> >> >> Am 11.02.2010 21:09, schrieb Olle E. Johansson: >>> >>> 11 feb 2010 kl. 13.30 skrev Klaus Darilion: >>> >>>> Am 11.02.2010 11:21, schrieb Armin Schindler: >>>>> Hello, >>>>> >>>>> 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. >>> >>>>> 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. >>>> >>>> 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. >>> >>> While we are discussing this, I would like some feedback. >>> >>> If we receive RTCP bye from the other end, we can close the port at that >>> point. >>> When we hang up the call, we send RTCP BYE and a final RTCP report. >>> >>> If we don't receive the RTCP BYE or a final report - I would like to keep >>> the RTCP port open a bit longer - but at maximum up to the destruction of >>> the SIP channel - so I can have a chance of receiving a final RTCP report >>> from the other end or/and RTCP BYE. >>> >>> What do you think? >> >> Will the channel only be kept alive in chan_sip or also in the core? Somehow >> we need a method to export the data received in the final reply, otherwise >> it makes no sense to wait. > > We have already come to the conclusion that there's no way to get this into > the CDRs, so in my pinefrog branch I'm sending the data over the manager > interface and storing it in a realtime storage facility - so it makes sense.
Then implement it! :-) klaus -- _____________________________________________________________________ -- 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
