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<tt>Hi Nick,<br>
<br>
It is risky to do a sync sleep as you keep the process blocked, so
the ability of OpenSIPS to handle new messages will decrease . <br>
<br>
Either do async sleep, either simple drop it to force a
retransmissions in 500ms - much simpler :)<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 02/07/2017 12:22 PM, Nick Altmann
wrote:<br>
</div>
<blockquote
cite="mid:CAJOe1EOMZg=Xmeq218k0t1g=+ydTRQ_cwZGvxk76suXe2OPMtg@mail.gmail.com"
type="cite">
<div dir="ltr">The easies way we use in such cases is:<br>
if (is_method("INVITE")) usleep("500");<br>
for in-dialog INVITEs (inside loose_route() / match_dialog()
section).<br>
<div class="gmail_extra"><br>
</div>
<div class="gmail_extra">It will delay slightly all reINVITES,
but also will guarantee correct order of packets.</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">2017-02-06 22:46 GMT+03:00 Stas
Kobzar <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:stas.kobzar@modulis.ca" target="_blank">stas.kobzar@modulis.ca</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">Hello Bogdan,
<div><br>
</div>
<div>In my case, ACK for previous INVITE has already
been received by OpenSIPS, but not sent yet. </div>
<div>In this case, will the variable $DLG_status still
equals 3 ?</div>
<div><br>
</div>
<div>Thanks</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote"><span class="">On Sun, Feb 5,
2017 at 11:15 AM, Bogdan-Andrei Iancu <span
dir="ltr"><<a moz-do-not-send="true"
href="mailto:bogdan@opensips.org"
target="_blank">bogdan@opensips.org</a>></span>
wrote:<br>
</span>
<div>
<div class="h5">
<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
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="m_9125869299888279809m_-8994335413096070854emphasis"><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 moz-do-not-send="true"
class="m_9125869299888279809m_-8994335413096070854moz-txt-link-freetext"
href="http://www.opensips.org/html/docs/modules/2.2.x/dialog.html#id297400"
target="_blank">http://www.opensips.org/html/d<wbr>ocs/modules/2.2.x/dialog.html#<wbr>id297400</a><br>
<br>
Regards,<br>
</tt>
<pre class="m_9125869299888279809m_-8994335413096070854moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" class="m_9125869299888279809m_-8994335413096070854moz-txt-link-freetext" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.<wbr>com</a></pre><span>
<div class="m_9125869299888279809m_-8994335413096070854moz-cite-prefix">On 02/02/2017 10:31 PM, Stas Kobzar
wrote:
</div>
</span><blockquote type="cite"><div><div class="m_9125869299888279809h5">
<div dir="ltr"><font face="monospace, monospace">Hello List,</font>
<div><font face="monospace, monospace">
</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">
</font></div>
<div><font face="monospace, monospace">UA
OpenSIPS PSTN GW</font></div>
<div><font face="monospace, monospace">------------------------------<wbr>-------------</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">
</font></div>
<div><font face="monospace, monospace">(UA sends ACK then new
INVITE)</font></div>
<div><font face="monospace, monospace">
</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">
</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">
</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">
</font></div>
<div><font face="monospace, monospace">My question is:
</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">
</font></div>
<div>
<div><font face="monospace, monospace">Thank you,</font></div>
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