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 <[email protected]>: > 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 <[email protected]>: > >> 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 -- >> 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 >> > > > > > > -- > --------------------------------------- > 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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