[asterisk-bugs] [JIRA] (ASTERISK-22795) SIP TLS calls stop working after a period of no SIP TLS calls to a destination

[Rusty Newton (JIRA)]

Rusty Newton created ASTERISK-22795:
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             Summary: SIP TLS calls stop working after a period of no SIP TLS calls to a destination
                 Key: ASTERISK-22795
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22795
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Channels/chan_sip/TCP-TLS
    Affects Versions: SVN, 1.8.23.1
         Environment: Asterisk 1.8.23.1
CentOS 6.4 x86_64
SIP TLS / SRTP
            Reporter: Dwayne Hubbard


SIP TLS/SRTP calls to a SIP TLS destination will setup a tcptls connection to the SIP TLS destination which is viewable using Asterisk CLI 'sip show tcp'.  Calls to a SIP TLS destination will work until there is a period (~30 minutes) of no activity to the SIP TLS destination at which point the tcptls _sip_tcp_helper_thread function will become blocked in the ast_poll() function with a timeout of -1.  Once this happens, SIP TLS calls to the SIP TLS destination will not succeed until one of the following occurs:

  1)  Asterisk Restarted
  2)  The chan_sip.so module is reloaded
  3)  A SSL_shutdown failed: 5 ERROR occurs

The patch provided change the _sip_tcp_helper_thread function timeout to 10 seconds.  If the ast_poll() function returns 0 (timeout) AND the tcptls AO2 reference count is greater than 2, then continue will be called to return to the ast_poll() function for another timeout period.  If the ast_poll() function returns 0 (timeout) AND the tcptls AO2 reference count is 2 (or less), then the tcptls session will be destroyed.

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