[asterisk-dev] Asterisk + WebRTC: No audio on any direction

[Matthew Jordan]

On Fri, Sep 4, 2015 at 7:51 PM,  <asandovalros at gmail.com> wrote:
> Hello everyone. I'd appreciate a lot your help with this issue. I'm running
> a very basic script of JS for subscribing my jsSIP User Agent to my local
> Asterisk server and making a voice call. I don't get any warnings or errors
> from the Asterisk CLI, but when I make a call to a legacy SIP phone or SIP
> trunk well configured, there is no audio on any side although there is
> ringing, calls can be answered and they never drop.
>
> The IP address of the SIP messages is correct both in the header of the
> message and in the RTP description, and it succeeds with sending ICE
> candidates. My Asterisk 12 was compiled with SRTP and pjproject. I don't get
> any error or warning messages on Asterisk, and I suppose that the SIP
> messages are ok.
>
> I read at the Asterisk WebRTC Wiki
> (https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support) this:
> "Starting with Asterisk 12 you need to have pjproject libraries installed,
> otherwise you most likely won't have audio in your WebRTC calls and no
> warning whatsoever!"
> I properly installed it and selected it for the Asterisk compilation, but I
> wonder wether I did it wrong, and how can I check it ...
>
> These are my files:
>
> http.conf
> [general]
> enabled=yes;
> bindaddr=0.0.0.0;
> bindport=8088;
> prefix=asterisk;
> tlsenable=yes;
> tlsbindaddr=0.0.0.0:8089;
> tlscertfile=/etc/asterisk/keys/asterisk.pem;
> tlsprivatekey=/etc/asterisk/keys/asterisk.pem;
>
> rtp.conf
> [general]
> rtpstart=10000;
> rtpend=20000;
> icesupport=true;
> stunaddr=stun.l.google.com:19302;
>
> sip.conf
> [general]
> context=toSipTrunk
> allow=ulaw
> allow=alaw
> allow=gsm
>
> [1000] ;legacy softphone (zoiper)
> secret=******
> type=friend
> host=dynamic
> dtmfmode=rfc2833
> disallow=all
> allow=ulaw
> allow=alaw
> context=myContext
>
> [1001] ;jsSIP User Agent
> type=friend
> username=1001
> host=dynamic
> secret=******
> encryption=yes
> avpf=yes
> icesupport=yes
> directmedia=no
> transport=udp,ws
> force_avp=yes
> dtlsenable=yes
> dtlsverify=no
> disallow=all
> allow=ilbc
> allow=g729
> allow=gsm
> allow=g723
> allow=ulaw
> dtlscertfile=/etc/asterisk/keys/asterisk.pem
> dtlsprivatekey=/etc/asterisk/keys/asterisk.pem
> dtlssetup=actpass
> context=myContext
>
> ... Thanks in advance

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-- 
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org