[asterisk-bugs] [JIRA] (ASTERISK-22750) SIP TLS calls stop working after a period of no SIP TLS calls to a destination
Rusty Newton (JIRA)
noreply at issues.asterisk.org
Thu Oct 31 09:44:03 CDT 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-22750?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Rusty Newton updated ASTERISK-22750:
------------------------------------
Status: Open (was: Triage)
> SIP TLS calls stop working after a period of no SIP TLS calls to a destination
> ------------------------------------------------------------------------------
>
> Key: ASTERISK-22750
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-22750
> 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
> Attachments: dw-asterisk-1.8.23.1-sip-tls.patch, dw-asterisk-trunk-r401662-sip-tls.patch
>
>
> 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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