[asterisk-bugs] [JIRA] (ASTERISK-24206) DTLS-SRTP fails on SIP over WebSockets call from SIPML5(chrome) to Asterisk

[Rusty Newton (JIRA)]

Rusty Newton created ASTERISK-24206:
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             Summary: DTLS-SRTP fails on SIP over WebSockets call from SIPML5(chrome) to Asterisk
                 Key: ASTERISK-24206
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24206
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Channels/chan_sip/General, Channels/chan_sip/WebSocket
    Affects Versions: SVN, 12.4.0
         Environment: Asterisk SVN-branch-12-r420805 (August 11th 2014), Chrome (38.0.2114.2 dev), Chrome (36.0.1985.125). SIPML5 live demo (?svn=224)
            Reporter: Rusty Newton


Attempting to make a call from SIPML5 in Chrome to a Playback of demo-congrats in Asterisk. Call fails upon hitting the playback.

{noformat}
  == Using SIP RTP CoS mark 5
    -- Executing [1000 at default:1] Answer("SIP/354-00000004", "") in new stack
    -- Executing [1000 at default:2] Playback("SIP/354-00000004", "demo-congrats") in new stack
    -- <SIP/354-00000004> Playing 'demo-congrats.gsm' (language 'en')
[Aug 11 16:28:52] ERROR[31257][C-00000004]: res_rtp_asterisk.c:1732 __rtp_recvfrom: DTLS failure occurred on RTP instance '0x7f6540009138' due to reason '(null)', terminating
[Aug 11 16:28:52] WARNING[31257][C-00000004]: res_rtp_asterisk.c:3944 ast_rtp_read: RTP Read error: Unspecified.  Hanging up.
[Aug 11 16:28:52] WARNING[31257][C-00000004]: app_playback.c:493 playback_exec: Playback failed on SIP/354-00000004 for demo-congrats
{noformat}

Once Asterisk hits the sound, we see a DTLS failure and the call disconnects.
Attached full debug file with SIP trace.

h2. Environment detail:

Asterisk SVN-branch-12-r420805 (August 11th 2014), Chrome (38.0.2114.2 dev), Chrome (36.0.1985.125). SIPML5 live demo (?svn=224) 

Machines involved:
 * Chrome(SIPML5) at 10.24.17.254
 * Asterisk at 10.24.18.124





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