[asterisk-bugs] [JIRA] (ASTERISK-26313) chan_sip : SDP problem changing Encryption yes to no

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26313?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=231941#comment-231941 ] 

Asterisk Team commented on ASTERISK-26313:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> chan_sip : SDP problem changing Encryption yes to no
> ----------------------------------------------------
>
>                 Key: ASTERISK-26313
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26313
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 13.10.0
>         Environment: Debian 8 3.16.0-4-amd64
>            Reporter: benasse
>            Severity: Minor
>         Attachments: full.log, sip1.conf, sip2.conf
>
>
> Below, the steps to reproduce :
> * I have a working peer, configured as in sip1.conf
> * I delete the parameters related to WebRTC as in sip2.conf
> * I do a SIP reload
> * When I try to make a call I get the following message in the asterisk CLI :
> {code}WARNING[5107][C-00000004] chan_sip.c: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio{code}
> Attached is the asterisk full log with the trace of the failed call with the SIP debug.
> callid asterisk : C-00000004
> CallID SIP : f51acc7c-7893-4f17-bde9-36865e50cf9f
> After a restart of asterisk, line working is properly.
> Do not hesitate if you want more information.



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