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. 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