[asterisk-bugs] [JIRA] (ASTERISK-28882) res_pjsip: Contact not completely removed on transport closure

[Asterisk Team (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-28882?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Asterisk Team updated ASTERISK-28882:
-------------------------------------


> res_pjsip: Contact not completely removed on transport closure
> --------------------------------------------------------------
>
>                 Key: ASTERISK-28882
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28882
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip
>    Affects Versions: 16.10.0
>         Environment: ubuntu 18.04 docker
> realtime jssip over WS
>            Reporter: Robert Sutton
>            Assignee: Asterisk Team
>            Severity: Minor
>         Attachments: asterisk.log.gz
>
>
> The logs are full of the below ERROR, they increase in frequency as time goes by.
> It appears that when a remote sip client disconnects abruptly (VPN disconnect) the contact is replaced but some part of the qualify code is still retaining the old contact id and trying to qualify it.
> As the code to timeout the contact has already removed it's reference to this dead contact it is never cleaned up.
> {noformat}
> ERROR[506]: res_pjsip.c:4278 endpt_send_request: Error 171060 'Unsupported transport (PJSIP_EUNSUPTRANSPORT)' sending OPTIONS request to endpoint 113
> {noformat}
> I've acquired the back trace for this error message...
> {noformat}
> # 0: [0x5db45d] asterisk utils.c:2404 __ast_assert_failed()
> # 1: [0x7fe8771b0ea8] res_pjsip.so res_pjsip.c:4285 endpt_send_request()
> # 2: [0x7fe8771b1625] res_pjsip.so res_pjsip.c:4477 ast_sip_send_out_of_dialog_request()
> # 3: [0x7fe8771b73ce] res_pjsip.so pjsip_options.c:908 sip_options_qualify_contact()
> # 4: [0x4645a7] asterisk astobj2_container.c:328 internal_ao2_traverse()
> # 5: [0x46487f] asterisk astobj2_container.c:414 __ao2_callback()
> # 6: [0x7fe8771b7561] res_pjsip.so pjsip_options.c:930 sip_options_qualify_aor()
> # 7: [0x7fe8771e671c] res_pjsip.so pjsip_scheduler.c:98 run_task()
> # 8: [0x5c5736] asterisk taskprocessor.c:1239 ast_taskprocessor_execute()
> # 9: [0x5cf3e0] asterisk threadpool.c:1354 execute_tasks()
> #10: [0x5c5736] asterisk taskprocessor.c:1239 ast_taskprocessor_execute()
> #11: [0x5cd02a] asterisk threadpool.c:367 threadpool_execute()
> #12: [0x5cec43] asterisk threadpool.c:1137 worker_active()
> #13: [0x5ce9dd] asterisk threadpool.c:1057 worker_start()
> #14: [0x5d83b7] asterisk utils.c:1249 dummy_start()
> #15: [0x7fe8c2e4e6ba] libpthread.so.0 :0 __pthread_get_minstack()
> #16: [0x7fe8c20e341d] libc.so.6 :0 clone()
> {noformat}
> here is the contact that "pjsip show contacts" lists
> {noformat}
>   Contact:  113/sip:na89q83p at 127.0.0.1:35636;transport=WS  f21857cece Avail        23.087
> {noformat}
> and here you can see it is qualifying 2 contacts, 
> {noformat}
> DEBUG[506]: res_pjsip/pjsip_options.c:857 sip_options_qualify_contact: Qualifying contact '113;@0f05565353fc038d2259556e7a9d87e1' on AOR '113'
> DEBUG[506]: res_pjsip/pjsip_options.c:857 sip_options_qualify_contact: Qualifying contact '113;@f21857cece07658a2e96e470283f3ea1' on AOR '113'
> {noformat}



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