<p dir="ltr">That makes a lot of sense.</p>
<p dir="ltr">Do you think is there anything I could do on my side to solve this? Maybe manually fix the RR?</p>
<p dir="ltr">This client has some kind of link balancing, so public IP may change at any time. This must be the reason he hasn't set it.</p>
<p dir="ltr">Thanks in advance!</p>
<div class="gmail_quote">Em 22/04/2016 7:40 PM, "Bogdan-Andrei Iancu" <<a href="mailto:bogdan@opensips.org">bogdan@opensips.org</a>> escreveu:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
<tt>Hi Daniel,<br>
<br>
Actually, on the A side, you have another proxy ( see the Record
Route with 172.20.17.11 in the INVITE). OpenSIPS tries to send the
BYE to the RR header, but that is private. A SIP proxy, if sending
traffic to public Internet, should not use at all private IPs.<br>
<br>
Bottom line, the broken link in your scenario is the </tt><tt>172.20.17.11
proxy before your opensips.<br>
<br>
Regards,<br>
</tt>
<pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre>
<div>On 23.04.2016 00:12, Daniel Zanutti
wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">Hi
<div><br>
</div>
<div>I'm facing an strange issue when my Opensips instance hangs
up a call, generating BYE to both sides (timeout on dialog
module or rtpproxy). The BYE is sent to both sides but A side
is behind NAT and the BYE is sent to the local IP address and
not to the public one.</div>
<div><br>
</div>
<div>See trace bellow:</div>
<div><br>
</div>
<div>Customer -> Opensips</div>
<div><br>
</div>
<div>
<div><b>U <a href="http://200.200.200.200:27923" target="_blank">200.200.200.200:27923</a>
-> <a href="http://199.199.199.199:5060" target="_blank">199.199.199.199:5060</a></b></div>
<div>INVITE <a href="mailto:sip%3A551133333333@plat.test.com" target="_blank">sip:551133333333@plat.test.com</a>
SIP/2.0</div>
<div>Record-Route:
<sip:172.20.17.11;lr;ftag=7db6f42e;did=769.5ee54854></div>
<div>Via: SIP/2.0/UDP
172.20.17.11:5060;branch=z9hG4bKb99a.3c0442d4.0</div>
<div>Via: SIP/2.0/UDP
172.28.0.12:57744;received=172.28.0.12;branch=z9hG4bK-524287-1---08559406a5e9137b;rport=57744</div>
<div>Max-Forwards: 68</div>
<div>Contact:
<a href="mailto:sip:100111@172.28.0.12:57744;rinstance=b4a1be0f56d73cfd" target="_blank"><sip:100111@172.28.0.12:57744;rinstance=b4a1be0f56d73cfd></a></div>
<div>To: <<a href="mailto:sip%3A551133333333@plat.test.com" target="_blank">sip:551133333333@plat.test.com</a>></div>
<div>From: <<a href="mailto:sip%3A100111@172.20.17.11" target="_blank">sip:100111@172.20.17.11</a>>;tag=7db6f42e</div>
<div>Call-ID: 79961NjFjMjY2MDhiYjhhNDYyN2UyMWZiMzFiNWM5NzBkNTM</div>
<div>CSeq: 2 INVITE</div>
<div>Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE,
REFER, INFO, OPTIONS, MESSAGE</div>
<div>Content-Type: application/sdp</div>
<div>Supported: replaces</div>
<div>User-Agent: X-Lite release 4.9.3 stamp 79961</div>
<div>Content-Length: 333</div>
<div>P-hint: NAT</div>
<div><br>
</div>
<div><br>
</div>
<div>BYE Opensips -> Customer</div>
<div><br>
</div>
<div><b>U <a href="http://199.199.199.199:5060" target="_blank">199.199.199.199:5060</a>
-> <font color="#ff0000"><a href="http://172.20.17.11:5060" target="_blank">172.20.17.11:5060</a></font></b></div>
<div>BYE
<a href="mailto:sip:100111@200.200.200.200:27923;rinstance=b4a1be0f56d73cfd" target="_blank">sip:100111@200.200.200.200:27923;rinstance=b4a1be0f56d73cfd</a>
SIP/2.0</div>
<div>Via: SIP/2.0/UDP
199.199.199.199:5060;branch=z9hG4bKe99a.e1c423a3.0</div>
<div>To: <<a href="mailto:sip%3A100111@172.20.17.11" target="_blank">sip:100111@172.20.17.11</a>>;tag=7db6f42e</div>
<div>From: <<a href="mailto:sip%3A551133333333@plat.test.com" target="_blank">sip:551133333333@plat.test.com</a>>;tag=as4088ffc9</div>
<div>CSeq: 1 BYE</div>
<div>Call-ID: 79961NjFjMjY2MDhiYjhhNDYyN2UyMWZiMzFiNWM5NzBkNTM</div>
<div>Route:
<sip:172.20.17.11;lr;ftag=7db6f42e;did=769.5ee54854></div>
<div>Max-Forwards: 70</div>
<div>Content-Length: 0</div>
<div>User-Agent: Softswitch</div>
</div>
<div><br>
</div>
<div><br>
</div>
<div>On the initial invite, I fixed the Contact using
fix_nated_contact() and signalling works fine between A and B
sides, the problem is happening when Opensips hangup the call,
because A side doesnt receive the BYE.</div>
<div><br>
</div>
<div>Do you guys have an idea on how to fix this? Maybe is it a
bug?</div>
<div><br>
</div>
<div>Thanks</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
</div>
<br>
<fieldset></fieldset>
<br>
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</pre>
</blockquote>
<br>
</div>
</blockquote></div>