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

[Alfonso Sandoval]

Thank you for the response. This is the first time that I require help on
these topics and I thought this list was the correct one. I've already sent
my subscription request to the asterisk-user mailing list, with the hope
that I find the answer to my problem soon.


Regards

On Mon, Sep 7, 2015 at 1:15 PM, Matthew Jordan <mjordan at digium.com> wrote:

> 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
>
> The asterisk-dev mailing list is for discussions regarding the actual
> source code of Asterisk. Please use the asterisk-users mailing list
> [1] for deployment, setup, troubleshooting, and other related
> questions.
>
> [1] http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> 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
>
> --
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