Dear navaismo,
thank you for your reply. Sorry for the delay but I was away for 2 weeks.
You were right. I have set another peer completely in sip_custom.conf and now I can set up the call (I see call in progress, In call, Call terminated). 
Unfortunately I have no audio. 
I have followed your tutorial and:
- My Asterisk has public IP address, and in peer I have nat=no. I can’t verify this configuration since sip show peers nor sip show peer doesn’t list NAT configuration.
In SIP debug I can see that “SIP read from WS:” and “c=IN IP4” point to the same client public IP address. - I have turn on RTP debug, and I see that Asterisk is sending RTP stream to the public IP address of client. Just I don’t see “(via ICE)”, and in SIP peer conf I have icesupport=yes and in rtp_custom.conf I have icesupport=true. Is this a problem? I em using Asterisk 11.17.1.
I have capture packets (tcpdump and Wireshark) on Asterisk and on client computer, and I can see that Asterisk is sending RTP stream to the client, and on client I see that it’s receiving RTP (UDP) stream.
It seams that client is getting the stream, just I don’t hear it in my Firefox.
How to continue from here?
Best regards.