[asterisk-bugs] [JIRA] (ASTERISK-25807) Asterisk & WebRTC with DTLS-SRTP

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25807?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=229701#comment-229701 ] 

Asterisk Team commented on ASTERISK-25807:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> Asterisk & WebRTC with DTLS-SRTP
> --------------------------------
>
>                 Key: ASTERISK-25807
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25807
>             Project: Asterisk
>          Issue Type: Information Request
>      Security Level: None
>    Affects Versions: 12.8.2
>         Environment: CentOS release 6.7 
>            Reporter: louk
>
> In short, I am creating a project based on WebRTC and Asterisk.
> - Asterisk 12.8.2
> - SipJS 0.7.3
> - Centos 6.7
> - Chrome 48.0.2
> - Self-signed certificate (for testing only)
> - Secure socket used between browser and Asterisk (wss://149.56.XX.XX:8089/ws)
> - Libsrtp & Pjproject installed
> - Asterisk server Directly connected to the Internet (Public IP)
> - Browser is behinf NAT
> - I must use DTLS-SRTP
> - WebRTC javascript code located in the same server as Asterisk. 
> - Asterisk IP: 149.56.XX.XX
> - Browser Public IP: 67.212.XX.XX
> - Browser Local IP: 192.168.0.115
> The signaling phase is correct, the peer is well connected to the server.
> The aim is to listen to a Playback or Saydigits from Asterisk server. When I run the call, I see that all goes well (SIP and RTP), but no sound in the browser (The volume is up).
> I searched on Google, in the forums, but no results .... is that it is a SRTP decryption problem? 
> Anyone tried this before?
> SIP.CONF
> -------------
> [1060] 
> type=friend
> username=1060 
> host=dynamic 
> secret=lookrtctest
> encryption=yes 
> avpf=yes
> icesupport=yes 
> context=outgoing
> directmedia=no
> transport=ws,wss
> force_avp=yes
> disallow=all
> allow=ulaw
> allow=alaw
> dtlsenable=yes
> dtlsverify=fingerprint
> dtlscertfile=/etc/asterisk/keys/asterisk.pem
> dtlscafile=/etc/asterisk/keys/ca.crt
> dtlssetup=actpass
> nat=yes,force_rport
> Extensions.conf:
> --------------------
> [outgoing]
> exten => _X.,1,Noop(*** Start Call *** )
> exten => _X.,n,Answer()
> exten => _X.,n,Playback(vm-from)
> exten => _X.,n,SayDigits(123456)
> exten => _X.,n,Hangup()
> RTP.conf:
> ------------
> [general]
> rtpstart=10000
> rtpend=20000
> icesupport=yes
> stunaddr=stun.l.google.com:19302
> Http.conf:
> ------------
> [general]
> enabled=yes
> bindaddr=0.0.0.0
> tlsenable=yes         
> tlsbindaddr=0.0.0.0:8089   
> tlsprivatekey=/etc/asterisk/keys/asterisk.pem
> tlscertfile=/etc/asterisk/keys/asterisk.pem
> RTP traces: 
> https://drive.google.com/file/d/0B9FF8D2noLW2R2F5TGdHLXBtWDg/view?usp=sharing
> Sip Traces: 
> https://drive.google.com/file/d/0B9FF8D2noLW2cU82S3htaVd1Q2M/view
> Wireshark traces:
> https://drive.google.com/file/d/0B9FF8D2noLW2bmZISGI1ZkUzV1U/view?usp=sharing



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