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

[Ian Gilmour (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-26003?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Ian Gilmour closed ASTERISK-26003.
----------------------------------

    Resolution: Not A Bug

> 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
>            Assignee: Unassigned
>
> 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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