[asterisk-bugs] [JIRA] (ASTERISK-26223) chan_sip: Changes to Encryption option not accepted on a reload of chan_sip

[Asterisk Team (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-26223?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Asterisk Team updated ASTERISK-26223:
-------------------------------------

    Assignee: Asterisk Team  (was: scgm11)
      Status: Triage  (was: Waiting for Feedback)

> chan_sip: Changes to Encryption option not accepted on a reload of chan_sip
> ---------------------------------------------------------------------------
>
>                 Key: ASTERISK-26223
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26223
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 13.9.1
>            Reporter: scgm11
>            Assignee: Asterisk Team
>            Severity: Minor
>         Attachments: full.log, sip1.conf, sip2.conf
>
>
> changing on the file and reloading sip Encryption from *yes* to *no* is not taking the change although "sip show channels" shows Encryption=no but this log is shown and the call is not possible, restarting asterisk make it work.
> chan_sip.c:10715 process_sdp: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio
> Added from a comment:
> 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.



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