[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:comment-tabpanel&focusedCommentId=240414#comment-240414 ] 

Joshua Colp commented on ASTERISK-27397:
----------------------------------------

I've reproduced this with the following dialplan logic:

{noformat}
exten => service,1,Answer
exten => service,n,Set(CONNECTEDLINE(name,i)=Boom)
exten => service,n,Set(CONNECTEDLINE(num)=500)
exten => service,n,Echo
{noformat}

This causes a connected line update to occur which results in a new key from Jitsi. This is seemingly not applied correctly as of Asterisk 14, and the SDP negotiation isn't quite right either in 15. Accepting this issue.

> 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: Unassigned
>         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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