Good configuration for early media

Asterisk Asterisk Support
2

No ideas ?
How do you configure your sip server for such architecture ?

I have one more question: I use qualify=yes but my first peer goes UNREACHABLE when the timeout is reach (qualifyfreq by default). The transport is udp.
When I debug I can see the OPTIONS between my server and the peers => the server try to contact the peer without using the private ip, is it normal ? We can see in the reply to the REGISTER than the private ip is used with ‘To:’.

This the debug:

<------------>
Scheduling destruction of SIP dialog '854708309' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:80.74.74.X:15120 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.98.131.Y:5060;branch=z9hG4bK6354bd17;rport=5060
From: "asterisk" <sip:asterisk@87.98.131.Y>;tag=as10f0d510
To: <sip:labo@192.168.48.132>;tag=618137457
Call-ID: 49d2d9662b26eacd4d2bd00340adc731@87.98.131.Y:5060
CSeq: 102 OPTIONS
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, SUBSCRIBE, NOTIFY, INFO
Accept: application/sdp
User-Agent: Media/3.3.99.11 (eXosip2/3.6.0)
Content-Length: 0

<------------->
...
<------------->

Sending to 80.74.74.X:15120 (NAT)
Reliably Transmitting (NAT) to 80.74.74.X:15120:
OPTIONS sip:labo@80.74.74.X:15120 SIP/2.0
Via: SIP/2.0/UDP 87.98.131.Y:5060;branch=z9hG4bK1aa721dd;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@87.98.131.Y>;tag=as153e898a
To: <sip:labo@80.74.74.X:15120>
Contact: <sip:asterisk@87.98.131.Y:5060>
Call-ID: 743129dc28dab8bb0897f36b1389b051@87.98.131.Y:5060
CSeq: 102 OPTIONS
User-Agent: Sip Server (20130523_12h10)
Date: Fri, 21 Mar 2014 10:26:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

Just tell me if sip.conf and extension.conf seems to be ok for such architecture. Thanks by advance.