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

[Joshua Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24205?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=221673#comment-221673 ] 

Joshua Colp commented on ASTERISK-24205:
----------------------------------------

Your analysis doesn't appear to be correct. While Asterisk may have initially attempted the public IP address for some reason it moved on to the ICE negotiation candidates when the ICE negotiation completed. The DTLS traffic was received by Chrome and it responded accordingly. Why the DTLS negotiation then failed I'm not sure.

> 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: 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
>         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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