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Hi Mohamed,<br>
<br>
Use t_relay(), the (transactional) statefull way to send out a SIP
request.<br>
<br>
Regards,<br>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="https://www.opensips-solutions.com">https://www.opensips-solutions.com</a>
<a class="moz-txt-link-freetext" href="https://www.siphub.com">https://www.siphub.com</a></pre>
<div class="moz-cite-prefix">On 31.07.2024 14:46, Mohamed OUALLA
wrote:<br>
</div>
<blockquote type="cite"
cite="mid:CAGmsPDvoKXVh55gKCEkOe_g3vbB6nSWAdrWt03thUCaievYgnA@mail.gmail.com">
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<div class="gmail_default"
style="font-family:monospace;font-size:large;color:#0b5394">[...]
<p> One last question: What is the most effective way to
connect the caller and callee in a simple VoIP call setup
(Caller - SIP Proxy - Callee)? Should I change the <b>$ru</b>
and then use <b>forward()</b> for stateless routing or <b>t_relay()</b>
for stateful routing, or is there another method you would
recommend?</p>
<p>Mohamed.</p>
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</div>
<br>
<div class="gmail_quote">
<div dir="ltr" class="gmail_attr">On Wed, Jul 31, 2024 at
4:00 AM Alex Balashov <<a
href="mailto:abalashov@evaristesys.com"
moz-do-not-send="true" class="moz-txt-link-freetext">abalashov@evaristesys.com</a>>
wrote:<br>
</div>
<blockquote class="gmail_quote"
style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">Hi,<br>
<br>
The Request URI is a SIP concept, while the destination URI
might be best described as a fictive invention of OpenSIPS
configuration script. It represents the next-hop destination
to which the request will be forwarded on the network and
transport layer, as you correctly surmised, while the request
URI is a logical destination. The destination URI supersedes
the request URI, but if the destination URI is not set, the
domain/port/transport attributes of the request URI are
consumed to determine the forwarding destination.<br>
<br>
An RURI is not the same thing as an Address of Record; an AoR
refers to a logical URI entity in the location service
(registrar) context. The purpose of a registrar is to map an
AoR (such as <a href="mailto:sip%3Amohamed@sip.opensips.org"
target="_blank" moz-do-not-send="true">sip:mohamed@sip.opensips.org</a>)
to one or more Contact URIs (e.g. <a
href="mailto:sip%3Aline1@192.168.1.100" target="_blank"
moz-do-not-send="true">sip:line1@192.168.1.100</a>;user=phone),
which indicate how to reach a given device on the network and
transport layer.<br>
<br>
Hopefully that helps!<br>
<br>
-- Alex<br>
<br>
> On Jul 30, 2024, at 9:26 PM, Mohamed OUALLA <<a
href="mailto:oualla.simohamed@gmail.com" target="_blank"
moz-do-not-send="true" class="moz-txt-link-freetext">oualla.simohamed@gmail.com</a>>
wrote:<br>
> <br>
> Hello all,<br>
> <br>
> I have a technical question about the difference
between Request URI and Destination URI in SIP. In my
understanding of SIP, the R-URI (Request URI) is located in
the start line of the SIP request and is also known as the
Address of Record (AoR). However, I am unclear about what the
Destination URI is for openSIPs. Is it the same as the Request
URI, or is it related to an added route header, or the
destination address in the transport protocol, I am not sure
about it?<br>
> <br>
> Additionally, I have observed that when I change the
$du pseudo variable in OpenSIPS, it relays the request to the
UAS without changing the R-URI (change it with the sip uri I
gave to $du pseudo variable). This behavior is the same as
using the t_relay() method, which also does not change the
R-URI but sends the request to the UAS. I guess that changes
have been done only for the destination address in the
transport layer.<br>
> <br>
> Could someone please explain these observations and
clarify the difference between R-URI and Destination URI?<br>
> And the best way to route calls from UAC to UAS in
simple VoIP call components (Caller - SIP Proxy - Callee),
actually I change the $ru, then I forward() the request
stateless or t_relay() stateful.<br>
> <br>
> Thank you.<br>
> _______________________________________________<br>
> Users mailing list<br>
> <a href="mailto:Users@lists.opensips.org"
target="_blank" moz-do-not-send="true"
class="moz-txt-link-freetext">Users@lists.opensips.org</a><br>
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class="moz-txt-link-freetext">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>
<br>
-- <br>
Alex Balashov<br>
Principal Consultant<br>
Evariste Systems LLC<br>
Web: <a href="https://evaristesys.com" rel="noreferrer"
target="_blank" moz-do-not-send="true"
class="moz-txt-link-freetext">https://evaristesys.com</a><br>
Tel: +1-706-510-6800<br>
<br>
<br>
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<span class="gmail_signature_prefix">-- </span><br>
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<div><span style="color:rgb(136,136,136)">==============================</span></div>
<div>Mohamed OUALLA</div>
<div><b><font color="#0b5394">VoIP Technical Solutions
and Software Engineer</font></b></div>
</div>
<div dir="ltr">Mail: <a
href="mailto:oualla.simohamed@gmail.com"
target="_blank" moz-do-not-send="true"
class="moz-txt-link-freetext">oualla.simohamed@gmail.com</a></div>
<div dir="ltr"><a
href="mailto:oualla.simohamed@gmail.com"
target="_blank" moz-do-not-send="true"><span
style="color:rgb(32,33,36)">N.Phone: +212 6 29 19
3116</span><br>
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style="text-align:center;background-color:rgb(255,255,255)"><b><font
size="4" face="monospace" color="#134f5c">SSC
Certified Professional</font></b></span></div>
<div dir="ltr"><span style="color:rgb(136,136,136)">==============================</span></div>
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