[asterisk-users] No audio on SIP over WebRTC

[Vinicius Fontes]

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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150727/5cf977cf/attachment.html>