| Summary: | ASTERISK-27482: SRTCP unprotect failed because of authentication failure | ||||
| Reporter: | Ahmet (hichkas) | Labels: | |||
| Date Opened: | 2017-12-14 07:55:01.000-0600 | Date Closed: | 2020-01-14 11:26:02.000-0600 | ||
| Priority: | Major | Regression? | |||
| Status: | Closed/Complete | Components: | |||
| Versions: | 13.18.3 | Frequency of Occurrence | Constant | ||
| Related Issues: |
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| Environment: | Asterisk 13.18.3 built by root @ raspbx on a armv6l running Linux Sip to webrtc video call | Attachments: | |||
| Description: | 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@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@from-internal:1] Macro("SIP/8001-0000001e", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/8001-0000001e", "1?theend") in new stack -- Goto (macro-hangupcall,s,3) -- Executing [s@macro-hangupcall:3] ExecIf("SIP/8001-0000001e", "0?Set(CDR(recordingfile)=)") in new stack -- Executing [s@macro-hangupcall:4] NoOp("SIP/8001-0000001e", "SIP/6002-0000001f monior file= ") in new stack -- Executing [s@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@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' | ||||
| Comments: | By: Asterisk Team (asteriskteam) 2017-12-14 07:55:02.192-0600 We appreciate the difficulties you are facing, however information request type issues would be better served in a different forum. The Asterisk community provides support over IRC, mailing lists, and forums as described at http://asterisk.org/community. The Asterisk issue tracker is used specifically to track issues concerning bugs and documentation errors. If this issue is actually a bug please use the Bug issue type instead. Please see the Asterisk Issue Guidelines [1] for instruction on the intended use of the Asterisk issue tracker. Thanks! [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines By: Asterisk Team (asteriskteam) 2017-12-14 07:55:03.212-0600 The severity of this issue has been automatically downgraded from "Blocker" to "Major". The "Blocker" severity is reserved for issues which have been determined to block the next release of Asterisk. This severity can only be set by privileged users. If this issue is deemed to block the next release it will be updated accordingly during the triage process. By: Asterisk Team (asteriskteam) 2017-12-14 07:55:03.352-0600 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]. | ||||