I am using an Asterisk 13.13.1 server behind a NAT firewall with FreePBX 126.96.36.199 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 188.8.131.52 37254
When I tested by making phone calls between the extensions I got the sip INVITE packets with STUN candidates as given below
Content-Length: 527 v=0 o=root 788094571 788094571 IN IP4 184.108.40.206 s=Asterisk PBX 13.13.1 c=IN IP4 220.127.116.11 t=0 0 m=audio 16136 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=ice-ufrag:54bf4e3c1e1db66768c03bb8427c02d1 a=ice-pwd:7ecf92c504f005395e9b50e20376e701 a=candidate:Hac1e002e 1 UDP 2130706431 10.10.10.10 16136 typ host a=candidate:S22c66e1f 1 UDP 1694498815 18.104.22.168 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 22.214.171.124 16137 typ srflx raddr 10.10.10.10 rport 16137 a=sendrecv
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?