[asterisk-users] SIP RTP ports not released when channel is hung up

[Marcus Hunger]

Hi,

did you see this one: https://issues.asterisk.org/view.php?id=16774 ? It
looks related to your issue.

Best regards, Marcus

On Fri, Feb 12, 2010 at 12:04 PM, Armin Schindler <armin at melware.de> wrote:

> On Fri, 12 Feb 2010, Armin Schindler wrote:
> >>>> 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 just made a test to confirm:
> After a restart of asterisk (to have a clean state with no sip channels
> activ and no RTP port allocated), I can confirm that:
> - REGISTER and OPTION listed sip channels don't use RTP ports
> - after some calls (e.g. SIP to SIP) the RTP ports are freed immediately
>   (looks like this is the case on hangup before answer).
> - after some other calls, the RTP ports are freed after about 20-30 seconds
>   after hangup.
> So basically all is correct.
>
> > 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.
>
> It really looks like that there is a case where the sip channel is not
> destroyed and that is the cause of the problem.
> I will try to reproduce this.
> Any ideas?
>
> Armin
>
>
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