Asterisk bypass NAT mobile operator

Asterisk Asterisk SIP
1

Hi there, I’ve a very complicated situation with my asterisk installation:

HOME SERVER:
IF1 > 192.168.1.2 (various services, asterisk port 5060udp. rtp 10002:10020)
        |   |
        |   |__>wan IF2 192.168.0.1 (natted from mobile ooperator)
        |
        |
        |______>tun0(vpn, 192.168.168.2)===TUNNEL===>VPS(Server with a static pubblic IP XX.XX.XX.XX. domain pp.ddns.com)



CLIENT VOIP:
MOBILE CONNECTION ----->VPS(pubblic IP XX.XX.XX.XX. domain pp.ddns.com)port 10161udp ----->tun0--->192.168.168.2(vpn HOME SERVER, port 5060)



I've tried to connect my mobile voip application with the scheme above, mobile connect, receive and send calls, but not audio (I think rtp problem).

Configurations asterisk (Asterisk 16.14.0) on Ubuntu 22.04:
 - sip.conf:
	[general]
	language=it
	tonezone=it
	keepalive=30
	pedantic=yes
	maxexpiry=3600
	rtptimeout=60
	rtupdate=yes
	rtpholdtimeout=180
	context=public
	allowoverlap=no
	udpbindaddr=0.0.0.0:5060
	transport=udp
	srvlookup=no
	disallow=all
	allow=ulaw
	allow=alaw
	allow=g722
	;=====PRESENCE
	callcounter=yes
	limitonpeers=yes
	notifyringing=yes
	notifyhold=yes
	notifycid=yes 
	allowsubscribe=yes
	;NAT
	nat=force_rport,comedia
	externip=IP_VPS
	media_address=pp.ddns.com
	externhost=pp.ddns.com
	localnet=192.168.0.0/255.255.0.0
	localnet=10.0.0.0/255.0.0.0
	localnet=172.16.0.0/255.240.0.0 
	localnet=169.254.0.0/255.255.0.0
	directmedia=no
	qualify=yes
	rtpkeepalive=5
 - rtp.conf:
    [general]
    rtpstart=10002
    rtpend=10020
    rtpchecksums=no
    strictrtp=no
    icesupport=no

Anyone help me to setup fine that situation?

Thanks

2

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