<div dir="ltr">Hello,<div><br><div>Finally, I came up with something that works as I want it.</div></div><div><br></div><div>I have tried to drop re-INVITE when DLG_lifetime is < 1sec. I worked. But, as the re-INVITEs were the part of direct media negotiation, I had some "cuts" in the audio in very beginning of the call.</div><div><br></div><div>So, I do following:</div><div>if ($DLG_lifetime < 2) usleep("200000");</div><div><br></div><div>This actually works fine for me.</div><div><br></div><div>Thank you for your help and ideas!</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Feb 7, 2017 at 8:45 AM, Stas Kobzar <span dir="ltr"><<a href="mailto:stas.kobzar@modulis.ca" target="_blank">stas.kobzar@modulis.ca</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi Bogdan,<div><br></div><div>It seems like $DLG_status will not work in my case.</div><div><br></div><div>But $DLG_lifetime probably can work. 5 seconds might be too long in my case, though. </div><div>I have re-INVITE almost immediately after initial INVITE (it is Asterisk, trying to renegotiate SDP media IP/port for direct media).<br></div><div><br></div><div>I guess 1 second might work fine here.</div><div><br></div><div>Thank you for your help!</div><div><br></div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Feb 7, 2017 at 5:10 AM, Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>></span> wrote:<br><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>
If the ACK git the loose_route() / match_dialog(), then the state
will move to 4. <br>
<br>
You may also use the $DLG_lifetime when handling the re-INVITE -
if it is less than 5 seconds, drop the re-INVITE :)<br>
<br>
Regards,<br>
</tt><span>
<pre class="m_2397412347747255373m_4288935020655814574moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="m_2397412347747255373m_4288935020655814574moz-txt-link-freetext" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.<wbr>com</a></pre>
</span><div><div class="m_2397412347747255373h5"><div class="m_2397412347747255373m_4288935020655814574moz-cite-prefix">On 02/06/2017 09:46 PM, Stas Kobzar
wrote:<br>
</div>
<blockquote type="cite">
<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">On Sun, Feb 5, 2017 at 11:15 AM,
Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@opensips.org" target="_blank"></a><a class="m_2397412347747255373m_4288935020655814574moz-txt-link-abbreviated" href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>></span> wrote:<br>
<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_2397412347747255373m_4288935020655814574m_-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 class="m_2397412347747255373m_4288935020655814574m_-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_2397412347747255373m_4288935020655814574m_-8994335413096070854moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="m_2397412347747255373m_4288935020655814574m_-8994335413096070854moz-txt-link-freetext" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.<wbr>com</a></pre><span>
<div class="m_2397412347747255373m_4288935020655814574m_-8994335413096070854moz-cite-prefix">On 02/02/2017 10:31 PM, Stas Kobzar
wrote:
</div>
</span><blockquote type="cite"><div><div class="m_2397412347747255373m_4288935020655814574h5">
<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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