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

[Armin Schindler]

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