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

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26003?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=230514#comment-230514 ] 

Asterisk Team commented on ASTERISK-26003:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> 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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