[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=222747#comment-222747 ] 

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

This issue also occurs for me when using PJSIP.

I've worked with a co-worker to at least track the issue to my Asterisk environment. Using a configuration that works for another system I still end up with the same failures as seen here, and yet a different failure if using wss instead of ws. I'll have to try with a completely fresh install on a new system/VM.

> 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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