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<tt>Hi Stas,<br>
<br>
Such races may happen at application level or even at network
level (when using UDP) - so if you have 2 packets very close as
time, they may swap. That is SIP :)<br>
<br>
The full guilt is in the UAC device, IMHO - it should let some
time gap between the ACK and re-INVITE, to eliminate any possible
races.<br>
<br>
Now, what you can do is to use the dialog module and to check the
dialog state when receiving the re-invite. If $DLG_status is </tt><tt><span
class="emphasis"><em>3</em></span> (Confirmed by a final reply
but no ACK received yet), drop with no reply the re-INVITEs (to
force a later retransmission) :<br>
<a class="moz-txt-link-freetext" href="http://www.opensips.org/html/docs/modules/2.2.x/dialog.html#id297400">http://www.opensips.org/html/docs/modules/2.2.x/dialog.html#id297400</a><br>
<br>
Regards,<br>
</tt>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
<div class="moz-cite-prefix">On 02/02/2017 10:31 PM, Stas Kobzar
wrote:<br>
</div>
<blockquote
cite="mid:CAMYcjopfMt-4kvjH58LhYBgJ9ADS3RQMic1xBkwyB8jpA4UtWg@mail.gmail.com"
type="cite">
<div dir="ltr"><font face="monospace, monospace">Hello List,</font>
<div><font face="monospace, monospace"><br>
</font></div>
<div><font face="monospace, monospace">My call flow has initial
INVITE and re-INVITE to update RTP IP/port.</font></div>
<div><font face="monospace, monospace">Usually everything works
well, but sometimes OpenSIPS come up with following example:</font></div>
<div><font face="monospace, monospace"><br>
</font></div>
<div><font face="monospace, monospace">UA
OpenSIPS PSTN GW</font></div>
<div><font face="monospace, monospace">-------------------------------------------</font></div>
<div><font face="monospace, monospace">INV(CSeq: 100) ----->
| ---> INV(CSeq: 100)</font></div>
<div><font face="monospace, monospace"><---- 200 OK
| <--- 200 OK</font></div>
<div><font face="monospace, monospace"><br>
</font></div>
<div><font face="monospace, monospace">(UA sends ACK then new
INVITE)</font></div>
<div><font face="monospace, monospace"><br>
</font></div>
<div><font face="monospace, monospace">ACK(CSeq: 100) ----->
|</font></div>
<div><font face="monospace, monospace">reINV(Cseq: 101) --->
|</font></div>
<div><font face="monospace, monospace"><br>
</font></div>
<div><font face="monospace, monospace">(OpenSIPS relays first
INVITE then ACK)</font></div>
<div><font face="monospace, monospace"> |
---> reINV(CSeq: 101)</font></div>
<div><font face="monospace, monospace"> |
---> ACK(CSeq: 100)</font></div>
<div><font face="monospace, monospace"><br>
</font></div>
<div><font face="monospace, monospace">When PSTN gateway
receives re-INVITE before ACK for previous INVITE</font></div>
<div><font face="monospace, monospace">it responds 500 with
Retry-After header. </font></div>
<div><font face="monospace, monospace">This is correct behaviour
which conforms to the RFC 3261 section 14.2</font></div>
<div><font face="monospace, monospace"><br>
</font></div>
<div><font face="monospace, monospace">My question is:<br>
</font></div>
<div><font face="monospace, monospace">Is it possible to assure
order of received and relayed messages within the same SIP
session? Is there any configuration parameter?</font></div>
<div><font face="monospace, monospace"> <br>
</font></div>
<div>
<div><font face="monospace, monospace">Thank you,</font></div>
-- <br>
<div class="gmail_signature" data-smartmail="gmail_signature">
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