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

Rusty Newton (JIRA) noreply at issues.asterisk.org
Thu Jun 25 08:44:33 CDT 2015


    [ https://issues.asterisk.org/jira/browse/ASTERISK-25065?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=226626#comment-226626 ] 

Rusty Newton commented on ASTERISK-25065:
-----------------------------------------

Okay, yeah that is a FreePBX version and not an Asterisk version.

FreePBX is a GUI and a distribution that includes Asterisk. Asterisk is the communications engine underneath.

Well, since you've upgraded and you can no longer reproduce the issue then I'm going to close this out. However it would be appreciated if you could figure out what Asterisk version you have upgraded from and to so that others will know where this issue exists and where it is possibly fixed.

Thanks!

> 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
>         Attachments: full, replication.txt
>
>
> 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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