[asterisk-dev] [svn-commits] twilson: trunk r275998 - in /trunk/channels: chan_sip.c sip/include/dialog.h

[Kevin P. Fleming]

On 07/14/2010 02:32 AM, Olle E. Johansson wrote:
> 
> 13 jul 2010 kl. 19.11 skrev SVN commits to the Digium repositories:
> 
>> Author: twilson
>> Date: Tue Jul 13 12:11:37 2010
>> New Revision: 275998
>>
>> URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=275998
>> Log:
>> Destroy RTP fds when we schedule final dialog destruction
>>
>> Since we are only keeping the dialog around for retransmissions at this point
>> and there is no possibility that we are still handling RTP, go ahead and
>> destroy the RTP sessions. Keeping them alive for 32 past when they are used
>> is unnecessary and can lead to problems with having too many open file
>> descriptors, etc.
>>
> After-commit reviewboard comment:
> 
> I don't agree fully with this change, Terry. There is still some RTCP activity that may happen after the BYE is sent or received. We should stay open for a few seconds, but possibly not 32 secs (or T1*64 as it really is). If we already have RTCP BYE, then it's fine to close the RTP. If not, we should propably hang around and wait for a few seconds of timeout, then close it. We will need those final RTCP reports for my code in pinefrog. If just close without receiving the final report from the other end (in combination with RTCP bye) vital statistics about QoS will be gone.

Good point Olle.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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