<div dir="ltr">did you turn on aggressive-nat-detection inside of sip_profiles/internal.xml what about NDLB, i have a box running that exact version without any issues regarding NAT<br></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Oct 16, 2014 at 3:40 AM, Miha <span dir="ltr"><<a href="mailto:miha@softnet.si" target="_blank">miha@softnet.si</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi,<br>
<br>
first my FS is on public ip.<br>
<br>
We have migrated from 1.2.x to Version<br>
1.4.8+git~20140910T181155Z~9f72b11892~64bit (git 9f72b11 2014-09-10<br>
18:11:55Z 64bit as we needed different server.<br>
<br>
I have noticed that a lot of costumers are complaining that a lot of<br>
phone call has been cut in about 1:30 min of the call. From what I can<br>
see from sip trace is that FS begins to send media to private ip.<br>
<br>
link to trace: <a href="http://pastebin.com/9M6X6SFp" target="_blank">http://pastebin.com/9M6X6SFp</a><br>
<br>
So, rtp is send first to right ip and media is ok but in about 1:30 min<br>
of the call the FS sends update and begin to send rtp to private ip.<br>
<br>
<br>
thx for help!<br>
<br>
miha<br>
<br>
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