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<div class="moz-cite-prefix">Hi Michael,<br>
<br>
you should use the following in your sip.conf<br>
nat = comedia,force_rport<br>
<br>
In this case, asterisk does not use the IP in the SDP and waits
for the stream and then responds to it. <br>
<br>
André<br>
<br>
On 16.10.2013 05:52, Michael Blake wrote:<br>
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<p class="MsoNormal">Hello list,</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">I have a configuration where I have 3 site
to site strongswan connections.</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">On one of my gateways I have an asterisk
server.</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">I am able to happily voip between my
asterisk servers and cisco call managers.</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">Now I am testing the roadwarrior case where
I have an android handset (Note II) running the android
strongswan client.</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">If I use any android voip client from one
of my remote sites and do not use the android strongswan
client, everything works.</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">When I set up a roadwarrior android/win7
connection from my android handset using the pk12 file I am
able to reach the web configuration page of the cisco call
manager behind the gateway.</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">However, the VOIP traffic goes from the
handset to the internal network, but the rtp packets do not
make it back to the android voip client (tried several
clients).</p>
<p class="MsoNormal">
</p>
<p class="MsoNormal">The SDP part of the SIP invite from the
android handset specifies my wifi router’s subnet, not the
virtual ip of the client.</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">From the sip invite (contact line shows
virtual ip)</p>
<p class="MsoNormal">….</p>
<p class="MsoNormal">Contact: <a moz-do-not-send="true"
href="sip:mabandroid@172.16.3.71:35469;ob">sip:mabandroid@172.16.3.71:35469;ob</a></p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">However the Sdp part shows the wifi address</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">c=IN IP4 192.168.10.106</p>
<p class="MsoNormal">a=rtcp:4001 IN IP4 192.168.10.106</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">When I turn on RTP debugging on the
asterisk gateway machine I see</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">Got RTP packet from <a
moz-do-not-send="true" href="http://172.16.3.71:4002">172.16.3.71:4002</a>
(type 00, seq 007834, ts 055840, len 000160)</p>
<p class="MsoNormal">Sent RTP packet to <a
moz-do-not-send="true" href="http://192.168.10.106:4002">192.168.10.106:4002</a>
(type 00, seq 065441, ts 055520, len 000160)</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">So packets coming from the virtual IP are
received from asterisk, but my voip client is listening for
UDP RTP packets on the wireless subnet as indicated. This
happened because of the SDP part of the message from the call
setup.</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">Phones I call from the android handset can
hear audio coming from the android handset, but the android
handset does not receive any audio.</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">
I have tried other voip clients that let you specify which
network to use (i.e. cellular, wifi,etc) but the strongswan
userspace network is not a selectable option.</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">I don’t have any knowledge of the potential
roadwarriors subnet so it seems an impossible scenario. I
assume that the default route gets used since the 10 subnet is
not configured anywhere (and would be impossible to predict
for a road warrior).</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">Does anyone have experience using the
android strongswan client and a voip android app
successfully? </p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">Michael</p>
<p class="MsoNormal">
</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal"> </p>
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