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

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25065?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=226154#comment-226154 ] 

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

Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. Please read over the Asterisk Issue Guidelines [1] which discusses the information necessary for your issue to be resolved and the format that information needs to be in. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need:

1. The specific steps or actions you took that caused you to encounter the problem.
2. The behavior you expected and the location of documentation that led you to that expectation.
3. The behavior you actually encountered.

To demonstrate the issue in detail, please include Asterisk log files generated per the instructions on the wiki [2]. If applicable, please ensure that protocol-level trace debugging is enabled, e.g., 'sip set debug on' if the issue involves chan_sip, and configuration information such as dialplan and channel configuration.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

[2] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information



> 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
>
> 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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