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Hi Brian,<br>
<br>
i think i found an issue. Please let me know if
aggressive-nat-detection will help as a do not find any appropriate
explenation what exactly does it do.<br>
<br>
Freeswitch in my case is for RTP, so opensips relays all call
throught FS and than FS sends it back to opensips which sends a call
to UAC.<br>
<br>
Problem is that Fs sends in the middle of the call UPDATE request
(which is ok) and than UAC sends back 200 ok. In this 200 ok the
contact address is private (also in SDP) as opensips does not
replace contact ip with public source ip , so FS begins to send
media to private ip which is the reason why media stops.<br>
<br>
I think this problem was also present with 1.2.x version but FS did
not sends updates. Do not understand this sentence wrong as FS is
not behaving wrong i am just explaning why I have this in new
version :)<br>
<br>
Will Aggressive nat detection fix this or I should fix this on
opensips and what exactly does aggresive nat detection do?<br>
<br>
tnx!<br>
<br>
miha<br>
<br>
<div class="moz-cite-prefix">On 16/10/2014 22:14, Chris Tunbridge
wrote:<br>
</div>
<blockquote
cite="mid:CAO3hyzGLNUmrSDx7QyTM_V9ONN7m4abjKxafxnPTQfC_isN7XA@mail.gmail.com"
type="cite">
<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 moz-do-not-send="true"
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 moz-do-not-send="true"
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>
_________________________________________________________________________<br>
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</blockquote>
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<br>
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<br>
<pre wrap="">_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
<a class="moz-txt-link-abbreviated" href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a>
<a class="moz-txt-link-freetext" href="http://www.freeswitchsolutions.com">http://www.freeswitchsolutions.com</a>
Official FreeSWITCH Sites
<a class="moz-txt-link-freetext" href="http://www.freeswitch.org">http://www.freeswitch.org</a>
<a class="moz-txt-link-freetext" href="http://confluence.freeswitch.org">http://confluence.freeswitch.org</a>
<a class="moz-txt-link-freetext" href="http://www.cluecon.com">http://www.cluecon.com</a>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
<a class="moz-txt-link-freetext" href="http://www.cudatel.com">http://www.cudatel.com</a>
FreeSWITCH-users mailing list
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<a class="moz-txt-link-freetext" href="http://www.freeswitch.org">http://www.freeswitch.org</a></pre>
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