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

[louk (JIRA)]

louk created ASTERISK-25807:
-------------------------------

             Summary: 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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