[asterisk-bugs] [JIRA] (ASTERISK-27482) SRTCP unprotect failed because of authentication failure

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27482?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=240627#comment-240627 ] 

Asterisk Team commented on ASTERISK-27482:
------------------------------------------

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].

> SRTCP unprotect failed because of authentication failure
> --------------------------------------------------------
>
>                 Key: ASTERISK-27482
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27482
>             Project: Asterisk
>          Issue Type: Information Request
>      Security Level: None
>    Affects Versions: 13.18.3
>         Environment: Asterisk 13.18.3 built by root @ raspbx on a armv6l running Linux
> Sip to webrtc video call
>            Reporter: Ahmet
>
> Sorry for bad English.
> When I call Webrtc sipml5 client from sip client audio is great video is shown but freezing 4-5 second, 1 second moving and freezing repeat.
> And I get only this error.
> == Using SIP VIDEO TOS bits 136
>   == Using SIP VIDEO CoS mark 6
>   == Using SIP RTP TOS bits 184
>   == Using SIP RTP CoS mark 5
>     -- Executing [6002 at from-internal:1] Dial("SIP/8001-0000001e", "SIP/6002") in new stack
>   == DTLS ECDH initialized (secp256r1), faster PFS enabled
>   == DTLS ECDH initialized (secp256r1), faster PFS enabled
>   == Using SIP VIDEO TOS bits 136
>   == Using SIP VIDEO CoS mark 6
>   == Using SIP RTP TOS bits 184
>   == Using SIP RTP CoS mark 5
>     -- Called SIP/6002
>     -- SIP/6002-0000001f is ringing
>   == SRTCP unprotect failed because of authentication failure
>     -- SIP/6002-0000001f answered SIP/8001-0000001e
>     -- Channel SIP/6002-0000001f joined 'simple_bridge' basic-bridge <e9c9c321-542a-456a-961b-feeb281cb833>
>     -- Channel SIP/8001-0000001e joined 'simple_bridge' basic-bridge <e9c9c321-542a-456a-961b-feeb281cb833>
>   == SRTCP unprotect failed because of authentication failure
>   == SRTCP unprotect failed because of authentication failure
>   == SRTCP unprotect failed because of authentication failure
>   == SRTCP unprotect failed because of authentication failure
>   == SRTCP unprotect failed because of authentication failure
>   == SRTCP unprotect failed because of authentication failure
>   == SRTCP unprotect failed because of authentication failure
>   == SRTCP unprotect failed because of authentication failure
>   == SRTCP unprotect failed because of authentication failure
>   == SRTCP unprotect failed because of authentication failure
>   == SRTCP unprotect failed because of authentication failure
>   == SRTCP unprotect failed because of authentication failure
>   == SRTCP unprotect failed because of authentication failure
>   == SRTCP unprotect failed because of authentication failure
>   == SRTCP unprotect failed because of authentication failure
>   == SRTCP unprotect failed because of authentication failure
>   == SRTCP unprotect failed because of authentication failure
>     -- Channel SIP/6002-0000001f left 'simple_bridge' basic-bridge <e9c9c321-542a-456a-961b-feeb281cb833>
>     -- Channel SIP/8001-0000001e left 'simple_bridge' basic-bridge <e9c9c321-542a-456a-961b-feeb281cb833>
>   == Spawn extension (from-internal, 6002, 1) exited non-zero on 'SIP/8001-0000001e'
>     -- Executing [h at from-internal:1] Macro("SIP/8001-0000001e", "hangupcall") in new stack
>     -- Executing [s at macro-hangupcall:1] GotoIf("SIP/8001-0000001e", "1?theend") in new stack
>     -- Goto (macro-hangupcall,s,3)
>     -- Executing [s at macro-hangupcall:3] ExecIf("SIP/8001-0000001e", "0?Set(CDR(recordingfile)=)") in new stack
>     -- Executing [s at macro-hangupcall:4] NoOp("SIP/8001-0000001e", "SIP/6002-0000001f monior file= ") in new stack
>     -- Executing [s at macro-hangupcall:5] AGI("SIP/8001-0000001e", "attendedtransfer-rec-restart.php,SIP/6002-0000001f,") in new stack
>     -- Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
>     -- <SIP/8001-0000001e>AGI Script attendedtransfer-rec-restart.php completed, returning 0
>     -- Executing [s at macro-hangupcall:6] Hangup("SIP/8001-0000001e", "") in new stack
>   == Spawn extension (macro-hangupcall, s, 6) exited non-zero on 'SIP/8001-0000001e' in macro 'hangupcall'
>   == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/8001-0000001e'



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