[asterisk-bugs] [JIRA] (ASTERISK-24205) DTLS-SRTP fails on SIP over WebSockets call from SIPML5(chrome) to Asterisk

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24205?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=224189#comment-224189 ] 

Rusty Newton commented on ASTERISK-24205:
-----------------------------------------

While running into this: ASTERISK-24639

I also ran into this issue again as well. It happens with the latest pjproject, Asterisk 13, openssl, etc and with pjsip instead of chan_sip.

{noformat}
    -- Executing [1000 at from-internal:2] Playback("Local/1000 at from-internal-00000000;2", "demo-congrats") in new stack
    -- <Local/1000 at from-internal-00000000;2> Playing 'demo-congrats.gsm' (language 'en')
    -- Call on PJSIP/101-00000000 placed on hold
    -- Started music on hold, class 'default', on channel 'Local/1000 at from-internal-00000000;1'
    -- PJSIP/101-00000000 answered Local/1000 at from-internal-00000000;1
    -- Stopped music on hold on Local/1000 at from-internal-00000000;1
    -- Channel Local/1000 at from-internal-00000000;1 joined 'simple_bridge' basic-bridge <134f1ae7-5ccd-4d01-917e-83120ed8e971>
    -- Channel PJSIP/101-00000000 joined 'simple_bridge' basic-bridge <134f1ae7-5ccd-4d01-917e-83120ed8e971>
[Dec 22 17:17:05] ERROR[6521][C-00000001]: res_rtp_asterisk.c:2040 __rtp_recvfrom: DTLS failure occurred on RTP instance '0x7f0be0010c18' due to reason '(null)', terminating
[Dec 22 17:17:05] WARNING[6521][C-00000001]: res_rtp_asterisk.c:4245 ast_rtp_read: RTP Read error: Unspecified.  Hanging up.
    -- Channel PJSIP/101-00000000 left 'simple_bridge' basic-bridge <134f1ae7-5ccd-4d01-917e-83120ed8e971>
    -- Channel Local/1000 at from-internal-00000000;1 left 'simple_bridge' basic-bridge <134f1ae7-5ccd-4d01-917e-83120ed8e971>
  == Spawn extension (from-internal, 101, 1) exited non-zero on 'Local/1000 at from-internal-00000000;1'
[Dec 22 17:17:05] WARNING[6504][C-00000000]: app_playback.c:493 playback_exec: Playback failed on Local/1000 at from-internal-00000000;2 for demo-congrats
{noformat}

Reproduction in this case occurred when originating between a Local channel dialing a playback and a Dial app dialing the SIPML5 client.

If the messages above do not appear, then we see this crash ASTERISK-24639

> DTLS-SRTP fails on SIP over WebSockets call from SIPML5(chrome) to Asterisk
> ---------------------------------------------------------------------------
>
>                 Key: ASTERISK-24205
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24205
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_rtp_asterisk
>    Affects Versions: SVN, 12.4.0, 13.1.0
>         Environment: Asterisk SVN-branch-12-r420805 (August 11th 2014), SVN-branch-13-r429983(Dec 22nd 2014, Chrome (38.0.2114.2 dev), Chrome (36.0.1985.125). Firefox 34.0 , SIPML5 live demo (?svn=224)
>            Reporter: Rusty Newton
>         Attachments: full_2.pcap, full_2.txt, full.txt, sip.conf.txt, sipDtls.conf
>
>
> 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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