Hi,

In the default configuration, the internal profile listen port 5060 and external profile listen port 5080.
And follow the setting in http://wiki.freeswitch.org/wiki/NAT_Traversal
vars.xml
<X-PRE-PROCESS cmd="set" data="bind_server_ip=59.120.xx.xx"/>
<X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/>

internal.xml
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>

external.xml
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>


When user make a SIP call from public network using external profile(5080), he encountered no audio.
Here is the SDP of 200 OK from FS to user.
v=0
o=FreeSWITCH 1366781127 1366781128 IN IP4 10.0.110.33
s=FreeSWITCH
c=IN IP4 10.0.110.33
t=0 0
m=audio 10568 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
m=video 10530 RTP/AVP 116
a=rtpmap:116 H264/90000
a=fmtp:116 profile-level-id=428014


When user make a SIP call from public network using internal profile(5060), everything OK.
Here is the SDP of 200 OK from FS to user.
v=0
o=FreeSWITCH 1366780920 1366780921 IN IP4 59.120.xx.xx
s=FreeSWITCH
c=IN IP4 59.120.xx.xx
t=0 0
m=audio 10892 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
m=video 10178 RTP/AVP 116
a=rtpmap:116 H264/90000
a=fmtp:116 profile-level-id=428014


I can't figure out which parameters in sofia profile or in dialplan make port 5060 set the public IP in SDP and work fine.
Any advice?

Thanks in advance.
Jimmy