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

[Ahmet (JIRA)]

Ahmet created ASTERISK-27482:
--------------------------------

             Summary: 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
            Severity: Blocker


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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