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

[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.

klaus