[asterisk-bugs] [JIRA] (ASTERISK-27397) res_srtp / res_pjsip_sdp_rtp: New key on answer to reinvite not applied correctly

[basildane (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27397?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=241148#comment-241148 ] 

basildane commented on ASTERISK-27397:
--------------------------------------

I ran a new test using Grandstream Wave in place of Jitsi.
The test was successful.

I'm going to open a ticket with Jitsi now.
Thanks for your insight into this problem.

> res_srtp / res_pjsip_sdp_rtp: New key on answer to reinvite not applied correctly
> ---------------------------------------------------------------------------------
>
>                 Key: ASTERISK-27397
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27397
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip_sdp_rtp, Resources/res_srtp
>    Affects Versions: 15.1.3
>         Environment: FreePBX distro 14
>            Reporter: basildane
>            Assignee: basildane
>              Labels: pjsip
>         Attachments: debug.log, pjsip.conf, pjsip.endpoint.conf
>
>
> Call between two SRTP endpoints work ok until transferred to a device that does not have SRTP.  Then Asterisk logs:
> {panel}
> res_srtp.c: SRTP unprotect failed with: authentication failure 110
> {panel}
> Audio is lost from the call at this point.



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