[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

[sst at sil.at]

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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/917/
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(Updated 2010-09-17 08:56:31.179212)


Review request for Asterisk Developers.


Changes
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have fix some code guideline things.
removed the dialog_unlink_needdestroy function and moved the unlink from this container into the dialog_unlink_all

i have tried to do the unlink and link of the rtptimeout in different places. the rtp_initialize function is only in 1.8 and this is for 1.6.2 but atleast if i place this into the create_addr_from_peer (which calls rtp_init) i just get the dialog of the second call leg into the container, not both (but i think we need both there). I also tried the sip_alloc function but there something with the ref count goes completly wrong.

another idea david?


Summary
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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 (updated)
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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