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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Have you tried turning ON the SIP ALG on the NS? This
should fix the contact header as it gets NATed on the way out.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>The Acme should be able to handle the NAT, we do that every day,
but everything on the Acme has to be set up for that, and they may not support
NAT traversal by choice. To troubleshoot why theirs isn’t would require seeing
the configuration and looking at any manipulations they have in place on the
relevant SIP Interfaces, realms, and local-policies.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>In my opinion, you are better off NOT depending on the provider’s
Acme to handle the traversal for you. It can work well with one phone at
a remote site, but for a whole PBX you are probably better off using the ALG in
your Netscreen or getting a SIP proxy for your end like a SIPerator or
Edgemarc. Also what would you do if the provider told you flat out they don’t
support NAT traversal? If all that fails you could always put a public on
the PBX, assuming the PBX vendor has the ability to secure itself.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> -Scott<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
voiceops-bounces@voiceops.org [mailto:voiceops-bounces@voiceops.org] <b>On
Behalf Of </b>milosz<br>
<b>Sent:</b> Friday, October 02, 2009 7:09 PM<br>
<b>To:</b> VoiceOps<br>
<b>Subject:</b> [VoiceOps] acme nat traversal clue<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>hi guys,<br>
<br>
need a bit of help... have never touched an acme before so i'm not sure what to
tell the people i am dealing with, and they seem incapable of resolving the
problem themselves. trying to explain the difference between layer 3 and
layer 7 is not working.<br>
<br>
setup: i have an ip pbx with a single interface at 10.0.1.80, and a netscreen
50 with a static routable ip natted to the pbx at 1.1.1.1 (sip alg is
off). my itsp has an acme at 2.2.2.2.<br>
<br>
this is what an inbound call looks like right now:<br>
<br>
2.2.2.2 -> 1.1.1.1 invite <a href="mailto:sip%3Axxxxxx@1.1.1.1">sip:xxxxxx@1.1.1.1</a><br>
10.0.1.80 -> 2.2.2.2 100 trying<br>
10.0.1.80 -> 2.2.2.2 200 ok (contact: <a href="mailto:sip%3Axxxxxx@10.0.1.80">sip:xxxxxx@10.0.1.80</a>)<br>
2.2.2.2 -> 10.0.1.80 ack <a href="mailto:sip%3Axxxxxx@10.0.1.80">sip:xxxxxx@10.0.1.80</a><br>
<br>
so obviously it breaks at this point.<br>
<br>
according to them, they are unable to configure the acme to respond to the
originating routable ip instead of the nonroutable ip in the sip contact.<br>
<br>
this is what someone at acme told them:<o:p></o:p></p>
<p><span style='font-size:11.0pt;color:#1F497D'>It’s not a configuration
issue, the endpoint at 1.1.1.1 has set it’s contact header address to
10.0.1.80 in it’s 200 OK response back to 2.2.2.2 (SBC). The SD will look
at the Contact header in order to determine where to send future in-dialog
requests such as ACK, BYE, Re-INVITEs etc (according to RFC3261). Please can you
check why 1.1.1.1 would be doing so?</span><o:p></o:p></p>
<p>this sounds irrelevant to me, since any sip endpoint will set its contact
header address to its own ip. sounds like this guy is saying "well
it's natted so that won't work, tell them to fix it." i certainly
can't configure my pbx to respond with a modified contact header.<o:p></o:p></p>
<p>can anyone provide some guidance here? or maybe i am just wrong and it
is impossible to make this work. but it looks like a basic hnt situation
to me.<o:p></o:p></p>
<p>thanks,<o:p></o:p></p>
<p>milosz<o:p></o:p></p>
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