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<font size="-1"><font face="Verdana">DISCLAIMER: I'm REALLY new to
FreeSwitch, so please take my advice with a grain of salt.<br>
<br>
I have a similar setup (and problem) - the wiki documentation refers to
it as "double nat". Like you, my FS and client are behind different
NATs and I can register my remote endpoint and make calls (in my case,
to the the FS demo ivr at 5000).<br>
<br>
Since your external endpoint (spa3102) is registering, you've
likely setup your sip profile correctly (ext-sip-ip, ext-rtp-ip, nat
settings, etc).<br>
<br>
According to the documentation, and my limited experience, you have at
least 2 options for NATted endpoints. However, I am unable to make the
second work.<br>
<br>
1) Setup stun on your remote endpoint (spa3102 in your case)<br>
2) Add <variable name="sip-force-contact"
value="NDLB-connectile-dysfunction"/> to the directory xml file that
describes your spa3102 endpoint<br>
<br>
Option 1 worked for me right away (eyebeam in my case) and, as
expected, the remote sdp had
the correct (remote) IP address, since the endpoint is using stun to
correctly identify its IP address to FS. However, option 2 has not made
a
difference (for me). Is it just me or is it strange that SIP works
without stun, but RTP doesn't?<br>
<br>
</font></font><font size="-1"><font face="Verdana">I guess I've been
spoiled by the way Asterisk handles NAT and was hopeful that </font></font><font
size="-1"><font face="Verdana">NDLB-connectile-dysfunction would
behave similarly, so I wouldn't have to tell users to setup stun on
their clients. </font></font><font size="-1"><font face="Verdana">Maybe
a FS user with some experience with this type of NAT setup and these
settings can help. I'd be interested in knowing how to correctly setup
remote NATted endpoints without stun - or, at least, hear from someone
that this setting works for them without stun.<br>
<br>
Anyway, hope this helps you with your SPA3102.<br>
<br>
</font></font><br>
Mark Campbell-Smith wrote:
<blockquote
cite="mid:33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com"
type="cite">
<pre wrap="">Hi!
I'm sure this is a NAT issue, but I'm not sure what options to use.
I have a Linksys SPA3102, NAT'd on the internet (remotely) and
connected to my FS on the otherside of the world, which is also
natted. A PAP2T is connected on the same subnet as the FS. The 3102
registers successfully and a call can be set up from the PAP2 to the
3102.
However, after FS receives the Remote SDP the audio stops (ring tone
stops in my case)
2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3646 Channel
sofia/internal/<a class="moz-txt-link-freetext" href="sip:2001@192.168.1.3:56885">sip:2001@192.168.1.3:56885</a> entering state
[completing][200]
2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3657 Remote SDP:
v=0
o=- 18490612 18490612 IN IP4 192.168.1.3
s=-
c=IN IP4 192.168.1.3
t=0 0
m=audio 16432 RTP/AVP 2 100 101
a=rtpmap:2 G726-32/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
I notice that the ip address in the o and c fields indicate a local IP
address. Should this IP address be an external IP address of the 3102
instead?
Thanks
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</pre>
</blockquote>
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