[asterisk-bugs] [JIRA] (ASTERISK-26544) res_rtp_asterisk: Delay in DTLS handshake causes audio setup delay

Mon Jan 9 14:12:10 CST 2017

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26544?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=234535#comment-234535 ] 

Marcelo Gornstein commented on ASTERISK-26544:
----------------------------------------------

Thank you guys, we are now running an environment where the SIP signalling (http.conf) is using our original certificate, while at the same time using the FreeSWITCH certificate for DTLS. Not ideal, but it works and perhaps this can be used in other scenarios where the same problem is present as a workaround. I'm not sure if this should be an issue to open to the OpenSSL guys or others.

In any case, thanks for your awesome help on this one!

Best regards

> res_rtp_asterisk: Delay in DTLS handshake causes audio setup delay
> ------------------------------------------------------------------
>
>                 Key: ASTERISK-26544
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26544
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_rtp_asterisk
>    Affects Versions: 13.6.0, 13.9.0, 13.12.0, 13.11.2, 13.12.1, 14.0.0, 14.1.1
>         Environment: Amazon Linux: 4.4.23-31.54.amzn1.x86_64 #1 SMP x86_64 GNU/Linux
> OpenSSL: 1.0.2j
> LibSRTP: 1.5.4
> SIPml5: e3152e1edf116b651de379b3cc971bf699787c26 (Fri Mar 4 09:47:48 2016 +0100)
> Chrome: 54.0.2840.71 (64-bit)
> FireFox: 49.0.2
> Opera: 41.0
> Online JSSip Demo at: https://tryit.jssip.net/
> Amazon EC2 instance
>            Reporter: Marcelo Gornstein
>            Assignee: Unassigned
>         Attachments: 54752_jira_asterisk_26544_v13_test.pcap, 54755_jira_asterisk_26544_v13_test_force_dtlsv1.pcap, cert.txt, dtls-handshake-audio-delay-asterisk-14.1.1-2.zip, dtls-handshake-audio-delay-asterisk-14.1.1-3.zip, dtls-handshake-audio-delay-asterisk-14.1.1.zip, freeswitch-bad.pcap, freeswitch-capture-call-without-delay.zip, freeswitch-ok.pcap, fs-dtls.crt, fs-dtls.key, jira_asterisk_26544_v14_force_dtlsv1.patch, jira_asterisk_26544_v14_test.patch, res_rtp_asterisk.c.rej, screencapture-sslanalyzer-comodoca-1482194006200.png, screencapture-ssllabs-ssltest-analyze-html-1480426789448.png, sip_trace.txt
>
>
> Hello,
> It seems that there is a delay in the audio setup when using WebRTC with latest Asterisk versions and latest browser versions (described in the Environment section).
> Sometimes there is no delay, but most of the time the delay goes between 1 second to a couple of minutes. 
> This seems to be related to a delay in the DTLS connection handshake between Asterisk and the browser (although this is just a guess after trying to isolate the issue).
> sip.conf
> {code}
> [100]
> nat=force_rport,comedia
> host=dynamic
> type=friend
> secret=secret
> disallow=all
> allow=g722
> icesupport=yes
> transport=wss
> dtlsenable=yes
> dtlsverify=no
> dtlscertfile=/cert.crt
> dtlsprivatekey=/cert.key
> dtlssetup=actpass
> videosupport=no
> encryption=yes
> avpf=yes
> force_avp=yes
> directmedia=no
> canreinvite=no
> context=wrtc
> {code}
> extensions.conf
> {code}
> [wrtc]
> exten => _X.,1,Answer
> same => n,Playback(tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys&tt-monkeys)
> same => n,Hangup
> {code}
> rtp.conf
> {code}
> [general]
> rtpstart=6000
> rtpend=65535
> icesupport=true
> [ice_host_candidates]
> x.x.x.x => y.y.y.y ; x.x.x.x is the internal IP, y.y.y.y is the external IP
> {code}
> [Edit by Rusty - removed excessive debug in description field as per issue tracker guidelines. Moving to attachment.]
> FireFox shows this in its logs:
> {code}
> 276467712[7f8929a4aec0]: [|WebrtcAudioSessionConduit] AudioConduit.cpp:612: GetAudioFrame 
> 276467712[7f8929a4aec0]: [|WebrtcAudioSessionConduit] AudioConduit.cpp:716: GetAudioFrame GetAudioFrame:Got samples: length 320 
> {code}
> But no audio is played until some (random) number of seconds pass, and this logs shows up:
> {code}
> 261443584[7f89265334a0]: Flow[568ce410feebc53d:0,rtp(none)]; Layer[dtls]: PacketReceived(2001)
> 261443584[7f89265334a0]: Flow[568ce410feebc53d:0,rtp(none)]; Layer[dtls]: Checking digest, algorithm=sha-256
> 261443584[7f89265334a0]: Flow[568ce410feebc53d:0,rtp(none)]; Layer[ice]:  SendPacket(75) succeeded
> 261443584[7f89265334a0]: Flow[568ce410feebc53d:0,rtp(none)]; Layer[dtls]: ****** SSL handshake completed ******
> 261443584[7f89265334a0]: Flow[568ce410feebc53d:0,rtp(none)]; Layer[dtls]: ALPN not negotiated, selecting default
> 261443584[7f89265334a0]: /builds/slave/m-rel-m64-00000000000000000000/build/src/media/mtransport/transportlayerdtls.cpp:865: Flow[568ce410feebc53d:0,rtp(none)]; 
> {code}
> EDIT: I forgot to mention that looking at the output from chrome://webrtc-internals, I noticed that during the delay the browser is stalled at ICEConnectionStateChecking. As soon as the audio is connected, it goes to ICEConnectionStateConnected.
>  
> Without changing any browser or network settings whatsoever either in the Asterisk Box or the browser's box, sometimes it just works. But most of the time the delay is present.
> With FreeSWITCH (cec0cb39830546a3a1c1df7ad7a05b05f14b8975 - Fri Oct 28 15:38:25 2016 -0500) works perfectly, every single time.
> Any help is greatly appreciated!
> Thank you.
> Best regards,

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