[asterisk-dev] [Code Review] Append two container for dialogs to delete and for rtp timeout checks to replace all dialogs container in ao2 callback for dialogs_needdestroy

[David Vossel]

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For some reason I was under the impression this patch was against Trunk instead of 1.6.2.

This is not entirely my call to make, but I do not believe this has a high probability of being accepted for merger into 1.8, and virtually no probability of being merged into 1.6.2.

If the patch is targeted for merger only into trunk, solving the RTP container issue is easy as we have already identified where linking should occur.  Were you expecting this to go into 1.6.2?  If so, I'd begin by discussing this with Russell.

- David


On 2010-09-17 08:56:31, schmidts wrote:
> 
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> (Updated 2010-09-17 08:56:31)
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> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> Instead of iterate through all sip dialogs in the do_monitor function of chan_sip.c i created two separate containers which links to dialogs. 
> The container dialogs_needdestroy links to dialogs which are marked as needdestroy, so they should be unrefered from the dialogs container.
> The container dialogs_rtpcheck links to dialogs which have active rtp, vrtp or trtp and rtptimeout, rtpholdtimeout or rtpkeepalive activated.
> 
> normaly the container with needdestroy dialogs is empty so the callback takes only around 30 usec. even if there are more than 10 dialogs to be destroyed it takes around 200 usec. 
> the callback for the rtpcheck tooks around 200 usec with 30 concurrent calls and 700 usec if the timeout is reached to do a softhangup.
> 
> further i have found in check_rtp_timeout a possible locking situation. i am not sure if this would be a working solution cause it could happen that a call would be hung up several ms or also 1 sec later.
> 
> i think after speeding the ao2 callback up, its not necessary to wait 1000 ms, with 100ms dialogs would be faster released, the timeout fires even shorter and scheduled events which has been scheduled in a very short future time would be fired faster. see change 40 for this.
> 
> i am also not sure if its necessary to iterate through the containers on module_unload to unlink the entries. the dialogs itself would be destroyed after they are unrefered from the dialogs container. see change 42 for this.
>  
> 
> 
> This addresses bug 17912.
>     https://issues.asterisk.org/view.php?id=17912
> 
> 
> Diffs
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> 
>   branches/1.6.2/channels/chan_sip.c 287304 
> 
> Diff: https://reviewboard.asterisk.org/r/917/diff
> 
> 
> Testing
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> 
> i have tested this with sipp and 500 concurrent calls all with rtptimeout 10 s and the active time was 10,9s per call.
> also the 20k peers register test run through without any problem. (with the astdbpatch)
> 
>                                  Messages  Retrans   Timeout   Unexpected-Msg
>       INVITE ---------->         500       0
>          100 <----------         500       0                   0
>          200 <----------         500       0                   0
>          ACK ---------->         500       0
>        Pause [    40.0s]         500                           500
>          BYE ---------->         0         0         0
>          200 <----------         0         0                   0
> ------------------------------ Test Terminated --------------------------------
> 
> 
> ----------------------------- Statistics Screen ------- [1-9]: Change Screen --
>   Start Time             | 2010-09-11 23:12:32
>   Last Reset Time        | 2010-09-11 23:12:47
>   Current Time           | 2010-09-11 23:12:48
> -------------------------+---------------------------+--------------------------
>   Counter Name           | Periodic value            | Cumulative value
> -------------------------+---------------------------+--------------------------
>   Elapsed Time           | 00:00:00:964              | 00:00:15:994
>   Call Rate              |    0.000 cps              |   31.262 cps
> -------------------------+---------------------------+--------------------------
>   Incoming call created  |        0                  |        0
>   OutGoing call created  |        0                  |      500
>   Total Call created     |                           |      500
>   Current Call           |        0                  |
> -------------------------+---------------------------+--------------------------
>   Successful call        |        0                  |        0
>   Failed call            |       35                  |      500
> -------------------------+---------------------------+--------------------------
>   Call Length            | 00:00:10:970              | 00:00:10:637
> ------------------------------ Test Terminated --------------------------------
> 
> 
> Thanks,
> 
> schmidts
> 
>