I am using an Asterisk 13.13.1 server behind a NAT firewall with FreePBX 18.104.22.168 and I have two chan_sip extensions.
And these extensions are also registered from behind NAT of different networks. I was able to register my extensions in Asterisk.
In order to successful voice communication over RTP, I noticed that it is essential to configure ICE for achieving my needs.
And I did the following steps
I have enabled ICE support for both extensions with FreePBX GUI, then configured STUN and TURN on SIP Settings.
And when I testing the stun status with asterisk CLI (using command: stun show status) , I got the following
Hostname Port Period Retries Status ExternAddr ExternPort
global.stun.twilio.com 3478 30 3 OK 22.214.171.124 37254
When I tested by making phone calls between the extensions I got the sip INVITE packets with STUN candidates as given below
o=root 788094571 788094571 IN IP4 126.96.36.199
s=Asterisk PBX 13.13.1
c=IN IP4 188.8.131.52
m=audio 16136 RTP/AVP 0 8 3 101
a=candidate:Hac1e002e 1 UDP 2130706431 10.10.10.10 16136 typ host
a=candidate:S22c66e1f 1 UDP 1694498815 184.108.40.206 16136 typ srflx raddr 10.10.10.10 rport 16136
a=candidate:Hac1e002e 2 UDP 2130706430 10.10.10.10 16137 typ host
a=candidate:S22c66e1f 2 UDP 1694498814 220.127.116.11 16137 typ srflx raddr 10.10.10.10 rport 16137
But the problem is that I don’t getting any candidates related with TURN. What would be the reason for this?
Do i need to setup any additional configurations?