[asterisk-users] webrtc no audio

[Vinicius Fontes]

I tested and it seems like I do have
https://issues.asterisk.org/jira/browse/ASTERISK-24146 but in a different
way. If I take more than 7s to answer the call, I don't get audio for a few
seconds (about 3), after that it works okay.



2015-08-28 10:43 GMT-03:00 Marek Červenka <cervajs at fpf.slu.cz>:

> 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
>
> *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>
>> 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>
>> http://www.api-digital.com --
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>
>
>
>
>
> --
> ---------------------------------------
> 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
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