[asterisk-bugs] [JIRA] (ASTERISK-27397) res_srtp / res_pjsip_sdp_rtp: New key on answer to reinvite not applied correctly
Joshua Colp (JIRA)
noreply at issues.asterisk.org
Tue Dec 5 12:32:08 CST 2017
[ https://issues.asterisk.org/jira/browse/ASTERISK-27397?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Joshua Colp updated ASTERISK-27397:
-----------------------------------
Component/s: Resources/res_pjsip_sdp_rtp
> 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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