<div dir="ltr"><div>Hello Jimmy,<br><br></div>Put the bind-server-ip to auto and set your public ip to ext-rtp-ip and ext-sip-ip in vars.xml file. Hope it will work.</div><div class="gmail_extra"><br clear="all"><div><b>Thanks<br>
=====================<br>
Manish Khilwani<br>
Skype: manish_khilwani<br>
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Phone : 919099923079</b><br></div>
<br><br><div class="gmail_quote">On Wed, Apr 24, 2013 at 4:29 PM, Jimmy Chang [via freeswitch-users] <span dir="ltr"><<a href="/user/SendEmail.jtp?type=node&node=7590072&i=2" target="_top" rel="nofollow" link="external">[hidden email]</a>></span> wrote:<br>
<blockquote style='border-left:2px solid #CCCCCC;padding:0 1em' class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class="HOEnZb"><div class="h5">
Hi,<br>
<br>
In the default configuration, the internal profile listen port 5060
and external profile listen port 5080.<br>
And follow the setting in
<a href="http://wiki.freeswitch.org/wiki/NAT_Traversal" rel="nofollow" link="external" target="_blank">http://wiki.freeswitch.org/wiki/NAT_Traversal</a><br>
vars.xml<br>
<X-PRE-PROCESS cmd="set" data="bind_server_ip=59.120.xx.xx"/><br>
<X-PRE-PROCESS cmd="set"
data="external_sip_ip=stun:<a href="http://stun.freeswitch.org" target="_blank" rel="nofollow" link="external">stun.freeswitch.org</a>"/><br>
<X-PRE-PROCESS cmd="set"
data="external_rtp_ip=stun:<a href="http://stun.freeswitch.org" target="_blank" rel="nofollow" link="external">stun.freeswitch.org</a>"/><br>
<br>
internal.xml<br>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/><br>
<br>
external.xml<br>
<param name="ext-sip-ip" value="$${external_sip_ip}"/><br>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/><br>
<br>
<br>
When user make a SIP call from public network using external
profile(5080), he encountered no audio.<br>
Here is the SDP of 200 OK from FS to user.<br>
<i>v=0</i><i><br>
</i><i>o=FreeSWITCH 1366781127 1366781128 IN IP4 10.0.110.33</i><i><br>
</i><i>s=FreeSWITCH</i><i><br>
</i><i>c=IN IP4 10.0.110.33</i><i><br>
</i><i>t=0 0</i><i><br>
</i><i>m=audio 10568 RTP/AVP 18 101</i><i><br>
</i><i>a=rtpmap:18 G729/8000</i><i><br>
</i><i>a=fmtp:18 annexb=no</i><i><br>
</i><i>a=rtpmap:101 telephone-event/8000</i><i><br>
</i><i>a=fmtp:101 0-16</i><i><br>
</i><i>a=silenceSupp:off - - - -</i><i><br>
</i><i>a=ptime:20</i><i><br>
</i><i>m=video 10530 RTP/AVP 116</i><i><br>
</i><i>a=rtpmap:116 H264/90000</i><i><br>
</i><i>a=fmtp:116 profile-level-id=428014</i><br>
<br>
<br>
When user make a SIP call from public network using internal
profile(5060), everything OK.<br>
Here is the SDP of 200 OK from FS to user.<br>
<i>v=0</i><i><br>
</i><i>o=FreeSWITCH 1366780920 1366780921 IN IP4 </i><i>59.120.xx.xx</i><i><br>
</i><i>s=FreeSWITCH</i><i><br>
</i><i>c=IN IP4 </i><i>59.120.xx.xx</i><i><br>
</i><i>t=0 0</i><i><br>
</i><i>m=audio 10892 RTP/AVP 18 101</i><i><br>
</i><i>a=rtpmap:18 G729/8000</i><i><br>
</i><i>a=fmtp:18 annexb=no</i><i><br>
</i><i>a=rtpmap:101 telephone-event/8000</i><i><br>
</i><i>a=fmtp:101 0-16</i><i><br>
</i><i>a=silenceSupp:off - - - -</i><i><br>
</i><i>a=ptime:20</i><i><br>
</i><i>m=video 10178 RTP/AVP 116</i><i><br>
</i><i>a=rtpmap:116 H264/90000</i><i><br>
</i><i>a=fmtp:116 profile-level-id=428014</i><i><br>
</i><br>
<br>
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.<br>
Any advice?<br>
<br>
Thanks in advance.<br>
Jimmy<br>
<br>
<br></div></div>_________________________________________________________________________
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