[asterisk-bugs] [JIRA] (ASTERISK-26003) Unexpected call termination when using PJSIP + SRTP

[Joshua Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26003?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=230518#comment-230518 ] 

Joshua Colp commented on ASTERISK-26003:
----------------------------------------

Thanks for the report and debug. However we also need protocol specific debug captured at the time of the issue. Please include the following:

* Asterisk log files generated using the instructions on the Asterisk wiki [1], with the appropriate protocol debug options enabled, e.g. 'pjsip set logger on' if the issue involves the chan_pjsip channel driver.
* Configuration information for the relevant channel driver, e.g. pjsip.conf.
* A wireshark compatible packet capture, captured at the same time as the Asterisk log output.

[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information



> Unexpected call termination when using PJSIP + SRTP
> ---------------------------------------------------
>
>                 Key: ASTERISK-26003
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26003
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 13.7.2
>         Environment: centos 6 (64-bit)
> asterisk 13.7.2
> pjproject 2.4.5
> libsrtp 1.5.2
>            Reporter: Ian Gilmour
>
> Hi, I'm seeing calls consistently terminating ~29-30mins into the call (say to the Asterisk echo()) when using chan_pjsip (with TLS) and SRTP (media_encryption=sdes). If I change pjsip.conf to use RTP (media_encryption=no) it all works fine.
> Should I be using another version of libsrtp? Or some other combination of source code packages?



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