[asterisk-users] webrtc no audio

[Vinicius Fontes]

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

*res_stun_monitor.conf:*
stunaddr = 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>:

> 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
> =======================================
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150827/ddb62e2e/attachment.html>