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? ---------- Forwarded message ---------- From: Vinicius Fontes <vinic...@aittelecom.com.br> Date: 2015-07-27 13:54 GMT-03:00 Subject: No audio on SIP over WebRTC To: Asterisk Users Mailing List - Non-Commercial Discussion < asterisk-users@lists.digium.com> I'm following this tutorial ( https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5) to deploy WebRTC support but I'm having an issue with RTP when the WebRTC softphone is behind NAT. In my scenario, the Asterisk server is running a public IPv4, and the softphone is behind NAT. I can register and make a call normally, but I don't get any audio in neither way (Asterisk/softphone and softphone/Asterisk). Using the very same config files but having the softphone and Asterisk on the same network it works fine. Any tips on how to solve this? Here's my relevant files. *;sip.conf:* [general] udpbindaddr=0.0.0.0:5060 realm=10.201.0.106 ;replace with your Asterisk server public IP address or host transport=udp,ws,wss tlsenable=yes tlsbindaddr=0.0.0.0 tlscertfile=/etc/asterisk/keys/asterisk.pem tlscafile=/etc/asterisk/keys/ca.crt tlscipher=ALL tlsclientmethod=tlsv1 [6000] host=dynamic secret=mysecret context=default type=friend icesupport=yes directmedia=no disallow=all allow=ulaw qualify=yes [6001] host=dynamic secret=mysecret context=default type=friend encryption=yes avpf=yes force_avp=yes icesupport=yes directmedia=no disallow=all allow=ulaw dtlsenable=yes dtlsverify=fingerprint dtlscertfile=/etc/asterisk/keys/asterisk.pem dtlscafile=/etc/asterisk/keys/ca.crt dtlssetup=actpass *extensions.conf:* [default] exten => _6XXX,1,Dial(SIP/${EXTEN}) *rtp.conf:* [general] rtpstart=10000 rtpend=20000 icesupport=yes stunaddr=stun.l.google.com:19302 2015-08-10 12:35 GMT-03:00 Joshua Colp <jc...@digium.com>: > Marek Cervenka wrote: > >> hello, >> >> i'm facing strange problem >> >> asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230 >> person1 to person3 are behind different NATs >> audio devices double checked >> >> call from person1(chrome) to person2(chrome) works >> call from person1(chrome) to person 3(chrome) - no audio on both side >> (RTP flowing only in one direction) >> call from person2(chrome) to person 3(chrome) - no audio on both side >> (RTP flowing only in one direction) >> BUT >> call from person2(chrome) to person 3(Jitsi sip client) - works! >> >> any tips howto find the problem? >> > > You would need to look at the ICE negotiation to see if it tried and > failed. After that would be looking at the DTLS negotiation. Asterisk > console output could provide some information. > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > > -- > _____________________________________________________________________ > -- 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 >
-- _____________________________________________________________________ -- 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