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. > 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.
If the channels exists even after 32 seconds after BYE, and BYE was signaled correctly, I would file a bug report. regards klaus > > Why aren't RTP ports released when not in use? > > Or is there a possibility to configure this behaviour? > > Thanks, > 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
