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Summary:ASTERISK-19565: Investigate failures of the nat_supertest in the Asterisk Test Suite
Reporter:Matt Jordan (mjordan)Labels:
Date Opened:2012-03-20 09:19:55Date Closed:
Priority:MajorRegression?No
Status:Open/NewComponents:Tests/testsuite
Versions:Frequency of
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Environment:Asterisk 1.8, 10, trunkAttachments:
Description:As part of the migration of the Asterisk Test Suite's management of SIPp processes over to twisted, it became clear that the SIPp scenarios that were part of the NAT supertest were failing.  This appears to be a problem with the test itself, as the SIPp scenarios themselves were never touched during the migration - and appeared to fail prior to the migration as well.

For every iteration of the test, we never receive a 100 in response to our REGISTER attempt:
{noformat}
------------------------------ Scenario Screen -------- [1-9]: Change Screen --
 Call-rate(length)   Port   Total-time  Total-calls  Remote-host
 10.0(0 ms)/1.000s   5061       1.10 s            1  127.0.0.1:5060(UDP)

 Call limit reached (-m 1), 0.000 s period  0 ms scheduler resolution
 0 calls (limit 30)                     Peak was 1 calls, after 0 s
 0 Running, 2 Paused, 0 Woken up
 0 dead call msg (discarded)            0 out-of-call msg (discarded)        
 1 open sockets                        

                                Messages  Retrans   Timeout   Unexpected-Msg
   REGISTER ---------->         1         0                            
        100 <----------         0         0         1         0        
        401 <----------         0         0         0         0        
------------------------------ Test Terminated --------------------------------


----------------------------- Statistics Screen ------- [1-9]: Change Screen --
 Start Time             | 2012-03-20 09:15:11:965 1332252911.965922            
 Last Reset Time        | 2012-03-20 09:15:13:072 1332252913.072173            
 Current Time           | 2012-03-20 09:15:13:072 1332252913.072423            
-------------------------+---------------------------+--------------------------
 Counter Name           | Periodic value            | Cumulative value
-------------------------+---------------------------+--------------------------
 Elapsed Time           | 00:00:00:000              | 00:00:01:106            
 Call Rate              |    0.000 cps              |    0.904 cps            
-------------------------+---------------------------+--------------------------
 Incoming call created  |        0                  |        0                
 OutGoing call created  |        0                  |        1                
 Total Call created     |                           |        1                
 Current Call           |        0                  |                          
-------------------------+---------------------------+--------------------------
 Successful call        |        0                  |        0                
 Failed call            |        0                  |        1                
-------------------------+---------------------------+--------------------------
 Call Length            | 00:00:00:000              | 00:00:01:001            
------------------------------ Test Terminated --------------------------------
{noformat}



{noformat}
[Mar 20 09:15:13] DEBUG[29123]: asterisk.sipp:153 __output_callback: Launching SIPp Scenario register.xml exited 1
[Mar 20 09:15:13] WARNING[29123]: asterisk.sipp:158 __output_callback: SIPp Scenario register.xml Failed [1, Resolving remote host '127.0.0.1'... Done.
2012-03-20 09:15:13:070 1332252913.070746: Call-Id: 1-29295@127.0.0.1, receive timeout on message register_client:1 without label to jump to (ontimeout attribute): aborting call.
]
{noformat}

This test needs to be investigated and fixed.
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