[asterisk-bugs] [JIRA] (ASTERISK-25065) SRTP failing over time

[Sam Ultima (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-25065?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Sam Ultima updated ASTERISK-25065:
----------------------------------

    Attachment: full
                sip_custom_post.conf
                sip_additional.conf
                sip.conf
                full.0
                fail2ban
                fail2ban.0
                extensions_override_freepbx.conf
                extensions_custom.conf
                extensions_additional.conf
                extensions.conf

Config & logfiles attached

> SRTP failing over time
> ----------------------
>
>                 Key: ASTERISK-25065
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25065
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/SRTP
>    Affects Versions: 13.3.2
>         Environment: Centos 6.5 (Current version: 6.12.65-27) running 4GB ram & dual CPU's. VOIP PHONES: Polycom IP670
>            Reporter: Sam Ultima
>            Assignee: Sam Ultima
>         Attachments: extensions_additional.conf, extensions.conf, extensions_custom.conf, extensions_override_freepbx.conf, fail2ban, fail2ban.0, full, full.0, New Text Document.txt, sip_additional.conf, sip.conf, sip_custom_post.conf
>
>
> From all appearances, SRTP appears to function except over time, calls begin failing. Temp solution is to restart phone and resume calls after registration.
> Debug shows the following errors:
> 2015-05-05 15:32:51] WARNING[3055][C-00000006] sdp_srtp.c: Unacceptable a=crypto tag: 13
> [2015-05-05 15:32:51] WARNING[3055][C-00000006] chan_sip.c: Rejecting secure audio stream without encryption details: audio 2228 RTP/SAVP 8 0 9 127
> SIP/2.0 488 Not acceptable here



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