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List: asterisk-dev
Subject: Re: [asterisk-dev] Asterisk + WebRTC: No audio on any direction
From: Alfonso Sandoval <asandovalros () gmail ! com>
Date: 2015-09-07 21:48:05
Message-ID: CAMhkM_kiOiJRx9m9g5YWgJ3kmC_vpw7pAsO_Z=X4wiakvXr-cA () mail ! gmail ! com
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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@digium.com> wrote:
> On Fri, Sep 4, 2015 at 7:51 PM, <asandovalros@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
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
[Attachment #5 (text/html)]
<div dir="ltr"><div>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. <br><br><br></div>Regards<br></div><div \
class="gmail_extra"><br><div class="gmail_quote">On Mon, Sep 7, 2015 at 1:15 PM, \
Matthew Jordan <span dir="ltr"><<a href="mailto:mjordan@digium.com" \
target="_blank">mjordan@digium.com</a>></span> wrote:<br><blockquote \
class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc \
solid;padding-left:1ex"><div class="HOEnZb"><div class="h5">On Fri, Sep 4, 2015 at \
7:51 PM, <<a href="mailto:asandovalros@gmail.com">asandovalros@gmail.com</a>> \
wrote:<br> > Hello everyone. I'd appreciate a lot your help with this issue. \
I'm running<br> > a very basic script of JS for subscribing my jsSIP User \
Agent to my local<br> > Asterisk server and making a voice call. I don't get \
any warnings or errors<br> > from the Asterisk CLI, but when I make a call to a \
legacy SIP phone or SIP<br> > trunk well configured, there is no audio on any side \
although there is<br> > ringing, calls can be answered and they never drop.<br>
><br>
> The IP address of the SIP messages is correct both in the header of the<br>
> message and in the RTP description, and it succeeds with sending ICE<br>
> candidates. My Asterisk 12 was compiled with SRTP and pjproject. I don't \
get<br> > any error or warning messages on Asterisk, and I suppose that the \
SIP<br> > messages are ok.<br>
><br>
> I read at the Asterisk WebRTC Wiki<br>
> (<a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support" \
rel="noreferrer" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support</a>) \
this:<br> > "Starting with Asterisk 12 you need to have pjproject libraries \
installed,<br> > otherwise you most likely won't have audio in your WebRTC \
calls and no<br> > warning whatsoever!"<br>
> I properly installed it and selected it for the Asterisk compilation, but I<br>
> wonder wether I did it wrong, and how can I check it ...<br>
><br>
> These are my files:<br>
><br>
> http.conf<br>
> [general]<br>
> enabled=yes;<br>
> bindaddr=0.0.0.0;<br>
> bindport=8088;<br>
> prefix=asterisk;<br>
> tlsenable=yes;<br>
> tlsbindaddr=<a href="http://0.0.0.0:8089" rel="noreferrer" \
target="_blank">0.0.0.0:8089</a>;<br> > \
tlscertfile=/etc/asterisk/keys/asterisk.pem;<br> > \
tlsprivatekey=/etc/asterisk/keys/asterisk.pem;<br> ><br>
> rtp.conf<br>
> [general]<br>
> rtpstart=10000;<br>
> rtpend=20000;<br>
> icesupport=true;<br>
> stunaddr=<a href="http://stun.l.google.com:19302" rel="noreferrer" \
target="_blank">stun.l.google.com:19302</a>;<br> ><br>
> sip.conf<br>
> [general]<br>
> context=toSipTrunk<br>
> allow=ulaw<br>
> allow=alaw<br>
> allow=gsm<br>
><br>
> [1000] ;legacy softphone (zoiper)<br>
> secret=******<br>
> type=friend<br>
> host=dynamic<br>
> dtmfmode=rfc2833<br>
> disallow=all<br>
> allow=ulaw<br>
> allow=alaw<br>
> context=myContext<br>
><br>
> [1001] ;jsSIP User Agent<br>
> type=friend<br>
> username=1001<br>
> host=dynamic<br>
> secret=******<br>
> encryption=yes<br>
> avpf=yes<br>
> icesupport=yes<br>
> directmedia=no<br>
> transport=udp,ws<br>
> force_avp=yes<br>
> dtlsenable=yes<br>
> dtlsverify=no<br>
> disallow=all<br>
> allow=ilbc<br>
> allow=g729<br>
> allow=gsm<br>
> allow=g723<br>
> allow=ulaw<br>
> dtlscertfile=/etc/asterisk/keys/asterisk.pem<br>
> dtlsprivatekey=/etc/asterisk/keys/asterisk.pem<br>
> dtlssetup=actpass<br>
> context=myContext<br>
><br>
> ... Thanks in advance<br>
<br>
</div></div>The asterisk-dev mailing list is for discussions regarding the actual<br>
source code of Asterisk. Please use the asterisk-users mailing list<br>
[1] for deployment, setup, troubleshooting, and other related<br>
questions.<br>
<br>
[1] <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" \
rel="noreferrer" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
<span class="HOEnZb"><font color="#888888"><br>
--<br>
Matthew Jordan<br>
Digium, Inc. | Director of Technology<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
Check us out at: <a href="http://digium.com" rel="noreferrer" \
target="_blank">http://digium.com</a> & <a href="http://asterisk.org" \
rel="noreferrer" target="_blank">http://asterisk.org</a><br> <br>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" \
rel="noreferrer" target="_blank">http://www.api-digital.com</a> --<br> <br>
asterisk-dev mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-dev" rel="noreferrer" \
target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-dev</a><br> \
</font></span></blockquote></div><br></div>
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-dev
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