[asterisk-users] webrtc no audio

[Marek Červenka]

are you sure you dont have this problem?
https://issues.asterisk.org/jira/browse/ASTERISK-24146

i'm now fighting with
https://issues.asterisk.org/jira/browse/ASTERISK-24602

Dne 27.8.2015 v 20:07 Vinicius Fontes napsal(a):
> I have it working now!
>
> *I had to install Asterisk 13 with PJSIP support.That's important, 
> even if you won't use PJSIP.* Then I did this configuration, which is 
> working fine under NAT:
>
> *sip.conf:*
> [6001]
> type=friend
> secret=REDACTED
> host=dynamic
> context=interno
> disallow=all
> ;allow=alaw,h263,h264,vp8
> allow=g722
> dtmf=auto
> videosupport=yes
> transport=ws,udp
> avpf=yes
> callerid="WebRTC" <6001>
> encryption=yes
> qualify=yes
> directmedia=no
> nat=force_rport,comedia
> icesupport=yes
> dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
> dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
> dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where 
> your DTLS cert file is
> dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where 
> your DTLS private key is
> dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when 
> setting up DTLS
>
> *rtp.conf:*
> icesupport=true
> stunaddr=stun.l.google.com:19302 <http://stun.l.google.com:19302>
>
> *res_stun_monitor.conf:*
> stunaddr = stun.l.google.com:19302 <http://stun.l.google.com:19302>   
>  ; Address of the STUN server to query.*
> *
> stunrefresh = 30
>
> 2015-08-12 5:23 GMT-03:00 Marek Červenka <cervajs at fpf.slu.cz 
> <mailto:cervajs at fpf.slu.cz>>:
>
>     Dne 11.8.2015 v 12:18 Joshua Colp napsal(a):
>
>         Vinicius Fontes wrote:
>
>             I'm having the same issue! The difference in my case is
>             Asterisk server
>             has a public IPv4 and the browser is behind a single NAT.
>
>             I'm forwarding my configuration below (which I posted
>             previously on
>             asterisk-users).
>
>             How can we debug ICE negotiation?
>
>
>         You have to do a packet capture, look at the exchange in
>         Wireshark, and see how the negotiation flows. It requires a
>         basic understanding of ICE.
>
>
>     it looks like we are facing this problem
>     https://issues.asterisk.org/jira/browse/ASTERISK-24146 too
>     if we use "[]" in sipml5 expert config "To disable TURN/STUN to
>     speedup ICE candidates gathering you can use an empty array. e.g. []."
>     it works better
>
>
>
>
>     -- 
>     ---------------------------------------
>     Marek Cervenka
>     =======================================
>
>
>     -- 
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>
>
>


-- 
---------------------------------------
Marek Cervenka
=======================================

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