I am still experiencing problem with lost media in conference on a client
behind NAT.
This is what I've done - disabled VAD on a NATed client and asked my friend
to produce lots of animal sounds in order to keep channel busy. But at the
end of minute sounds of wild nature disapeared again. We reproduced that
without security with tcp SIP transport and got the same result.
Then I started to dig into SIP trace and this is what I found.
This client (behind NAT) recieve subsequent INVITE message from FS which
seem to destroy dialog and causes client app to close media stream after a
session being established normally. I performed the same call from box with
public ip and saw no subsequent INVITE's from FS. How come FS sends an
INVITE message to already connected client? Is it OK? Should client handle
this normally?

Below is client's SIP trace:

INVITE sip:1.conference...@74.208.167.44:5081;transport=TLS SIP/2.0
...
User-Agent: DoxWox SIP user agent
..

SIP/2.0 100 Trying
..
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
..

SIP/2.0 407 Proxy Authentication Required
..
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
..

ACK sip:1.conference...@74.208.167.44:5081;transport=TLS SIP/2.0
..

SIP/2.0 100 Trying
..
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
..

SIP/2.0 183 Session Progress
..
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
..

SIP/2.0 200 OK
..
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
..

ACK sip:1.conference...@74.208.167.44:5081;transport=tls SIP/2.0
..

*Finally, this message cause media stream closing*
INVITE sip:1...@87.184.52.45:64183;transport=tls SIP/2.0
Via: SIP/2.0/TLS 74.208.167.44:5081;branch=z9hG4bK8269NDyXQNjyH
Max-Forwards: 70
From: 
<sip:1.conference...@74.208.167.44<sip%3a1.conference...@74.208.167.44>>;tag=vQH234QtN2U8Q

To: <sip:1...@74.208.167.44
<sip%3a1...@74.208.167.44>>;tag=3a231ba86c894ceca81d5021b68d3b6c

Call-ID: 37edc38329f64fe98c36cc0a6ddcbd9d
CSeq: 121093810 INVITE
Contact: <sip:1.conference...@74.208.167.44:5081;transport=tls>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Session-Expires: 120;refresher=uac
Min-SE: 120
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 340
v=0
o=FreeSWITCH 1254396503 1254396504 IN IP4 74.208.167.44
s=FreeSWITCH
c=IN IP4 74.208.167.44
t=0 0
m=audio 27726 RTP/SAVP 103 101
a=rtpmap:103 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:J6x2UgRVUY8GfbwjCuyyttrtnXnwwuWA9Pt+o3VW
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