[Home]

Summary:ASTERISK-27027: chan_sip: Channel Stuck when load reach to 500+
Reporter:Bhavik (RomanDcoz)Labels:
Date Opened:2017-06-01 01:59:27Date Closed:2020-01-14 11:13:54.000-0600
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:13.13.1 13.15.0 Frequency of
Occurrence		Frequent
Related
Issues:
Environment:operating system : Centos 7 64 Bit [root@localhost ~]# lscpu Architecture: x86_64 CPU op-mode(s): 32-bit, 64-bit Byte Order: Little Endian CPU(s): 1 On-line CPU(s) list: 0 Thread(s) per core: 1 Core(s) per socket: 1 Socket(s): 1 NUMA node(s): 1 Vendor ID: GenuineIntel CPU family: 6 Model: 94 Model name: Intel(R) Xeon(R) CPU E3-1230 v5 @ 3.40GHz Stepping: 3 CPU MHz: 3407.463 BogoMIPS: 6816.00 Hypervisor vendor: VMware Virtualization type: full L1d cache: 32K L1i cache: 32K L2 cache: 256K L3 cache: 8192K NUMA node0 CPU(s): 0 RAM : 50GB Hard Disk : 800GBAttachments:( 0) backtrace-threads.txt
( 1) core-show-locks.txt
Description:When we increase load 300 calls to 500 calls , call will stuck around 50%.
We have created a simple Dialplan like.
[IVR]
exten =>_X.,1,NoOp(IVR Start)
exten =>_X.,n,Answer()
exten =>_X.,n,Wait(1)
exten =>_X.,n,Agi(ivr.php,1)

I was added h extension but i tried with remove h extension as well.
In my ivr.php File just have getting  channel variables and playing an audio file.Thats it.

We have tested like send call via WINSIP dialer to load 500 call to specifc number and playing audio file vai AGI file.
after couple of minutes, we hangup all calls from WINSIP Dialer Side,around 50% call drop correctly but remaining calls are stuck and showing last message like : Rx: BYE
but channel showing still active.

Also every 20 sec around getting below warning as well.

[May 30 05:53:07] WARNING[24117]: chan_sip.c:4335 __sip_autodestruct: Autodestruct on dialog '70b2-09686995-0001-Call898' with owner SIP/XXXXXXX in place (Method: BYE). Rescheduling destruction for 10000 ms
[May 30 05:53:07] WARNING[24117]: chan_sip.c:4335 __sip_autodestruct: Autodestruct on dialog '143c-09566665-0001-Call418' with owner SIP/XXXXX in place (Method: BYE). Rescheduling destruction for 10000 ms


After call stuck, it


Comments:By: Asterisk Team (asteriskteam) 2017-06-01 01:59:28.562-0500

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].
By: Joshua C. Colp (jcolp) 2017-06-02 11:06:19.792-0500

We suspect that you have a deadlock occurring within Asterisk. Please follow the instructions on the wiki [1] for obtaining debug relevant to a deadlock. Once you have that information, attach it to the issue. Be sure the instructions are followed exactly as the debug may otherwise not be useful.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-GettingInformationForADeadlock


By: Bhavik (RomanDcoz) 2017-06-13 00:18:45.429-0500

This is back trace logs.
By: Bhavik (RomanDcoz) 2017-06-13 00:19:16.635-0500

This is lock file.
By: Bhavik (RomanDcoz) 2017-06-13 00:19:36.915-0500

I have created backtrace using above link, but not showing any lock after get a back trace.

Please find attached log files.

Also let me know if you require anything from my side.
By: Rusty Newton (rnewton) 2017-06-22 17:40:11.241-0500

It sounds like you are developing or testing an implementation, is there any reason you are using chan_sip (the old SIP driver) instead of the newer, core supported res_pjsip/chan_pjsip ?

I'm not sure we could make progress on this issue unless we can reproduce the problem. Since chan_sip isn't core supported, that isn't likely to happen very soon.

At the moment I don't have enough to go on here.

If you only run into the issue with chan_sip then you will want to post an entire Asterisk configuration that is trimmed down to only reproduce the issue along with all config files and scripts needed to reproduce. That way if someone does want to take the issue on they'll have everything they need to reproduce it.
By: Asterisk Team (asteriskteam) 2017-07-07 12:00:01.321-0500

Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1].

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines