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 <vinicius at 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 at 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 <jcolp at 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
>
>
> --
> _____________________________________________________________________
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