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

[Joshua Colp (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-27397?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua Colp updated ASTERISK-27397:
-----------------------------------

    Assignee: basildane  (was: Unassigned)
      Status: Waiting for Feedback  (was: Open)

[~basildane] I've done further testing as I thought the new key wasn't getting applied, but after further investigation it appears it is - and that is what breaks Jitsi. It does not appear to be applying its own new key to its outgoing media. It only applies it to RTCP. If I modify the code so that the new key isn't applied on the Asterisk side either then RTP works and I hear audio but RTCP fails to unprotect. This appears to be a bug in Jitsi.

Can you please provide another device that also exhibits the same issue and provide a sip trace as well for it?

> 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
>         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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