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[cervajs at fpf.slu.cz]

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?

--
---------------------------------------
Marek Cervenka
=======================================


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[jcolp at digium.com]

Marek Cervenka wrote:

[vinicius at aittelecom...]

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@aittelecom.com.br (vinicius@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 (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 <jcolp@digium.com (jcolp@digium.com)>:

[jcolp at digium.com]

Vinicius Fontes wrote:

[cervajs at fpf.slu.cz]

Dne 11.8.2015 v 12:18 Joshua Colp napsal(a):