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<tt>Hi Sanjeev,<br>
<br>
Well, things are getting a bit more complex here as you run
OpenSIPS behind a NAT (in a private network) - this means that
opensips will be 'visible' with different IPs by the parties in
the same private network and by the parties in the public network.<br>
<br>
In the same time you need to ensure that RTP is able to be routed
(at IP level) between the 2 endpoints - probably that's your
issue, that the private IP advertised in SDP by the end point in
the private network is not routable from the perspective of the
other end point in the public network -> the public end point
cannot send RTP to the private endpoint.<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>
OpenSIPS Bootcamp 2018
<a class="moz-txt-link-freetext" href="http://opensips.org/training/OpenSIPS_Bootcamp_2018/">http://opensips.org/training/OpenSIPS_Bootcamp_2018/</a>
</pre>
<div class="moz-cite-prefix">On 08/03/2018 10:10 AM, Sanjeev Sharma
wrote:<br>
</div>
<blockquote type="cite"
cite="mid:1147292412.1512420.1533280228500@mail.yahoo.com">
<div style="font-family:verdana, helvetica,
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<div style="font-family:verdana, helvetica,
sans-serif;font-size:13px;">
<div>Hi ALL</div>
<div><br>
</div>
<div>as a further dig into the media issue no voice between
caller and callee and found that the in the request packet
of invite header the value of via header is coming
incorrect ( as in the below packet is coming the local ip
address of the machine 192.1682.248 instead of <span><span
style="color: rgb(0, 0, 0); font-family: verdana,
helvetica, sans-serif;">159.200.37.234. </span></span></div>
<div><span><span style="color: rgb(0, 0, 0); font-family:
verdana, helvetica, sans-serif;"><br>
</span></span></div>
<div><span><span style="color: rgb(0, 0, 0); font-family:
verdana, helvetica, sans-serif;">my machine is behind
the fortigate firewall. could you please suggest how to
set or change the value of via header , is this will be
done at firewall level or to change the any value in the
opensip configuration file. below is sample packet </span></span></div>
<div><br>
</div>
<div><span>
<div>U 2018/08/03 16:27:25.156824 159.200.37.234:5060
-> 149.355.453.253:5060<br>
</div>
<div>INVITE <a class="moz-txt-link-freetext" href="sip:124007@149.355.453.253;transport=UDP">sip:124007@149.355.453.253;transport=UDP</a>
SIP/2.0.</div>
<div>Via: SIP/2.0/UDP
192.168.2.248:5060;branch=z9hG4bK-524287-1---81e029059dd869e2;rport.</div>
<div>Max-Forwards: 70.</div>
<div>Contact:
<a class="moz-txt-link-rfc2396E" href="sip:124010@159.200.37.234:5060;transport=UDP"><sip:124010@159.200.37.234:5060;transport=UDP></a>.</div>
<div>To: <a class="moz-txt-link-rfc2396E" href="sip:124007@149.355.453.253;transport=UDP"><sip:124007@149.355.453.253;transport=UDP></a>.</div>
<div>From:
<a class="moz-txt-link-rfc2396E" href="sip:124010@149.355.453.253;transport=UDP"><sip:124010@149.355.453.253;transport=UDP></a>;tag=29295014.</div>
<div>Call-ID: 0PqYYFAWxnuGearNQ73LxQ...</div>
<div>CSeq: 1 INVITE.</div>
<div>Content-Type: application/sdp.</div>
<div>User-Agent: Z 3.15.40006 rv2.8.20.</div>
<div>Allow-Events: presence, kpml, talk.</div>
<div>Content-Length: 241.</div>
</span><br>
</div>
<div><br>
</div>
<div>Thanks in advance </div>
<div>Sanjeev!!</div>
</div>
<div id="yahoo_quoted_3876342207" class="yahoo_quoted">
<div style="font-family:'Helvetica Neue', Helvetica, Arial,
sans-serif;font-size:13px;color:#26282a;">
<div> On Wednesday, 1 August, 2018, 5:41:02 PM GMT+10,
Sanjeev Sharma <a class="moz-txt-link-rfc2396E" href="mailto:sanjeevt510@yahoo.com"><sanjeevt510@yahoo.com></a> wrote: </div>
<div><br>
</div>
<div><br>
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<div style="font-family:verdana, helvetica,
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<div><span></span>
<pre class="yiv7416488403ydp632f2846yiv4421722051moz-signature" style="white-space:pre-wrap;">Hi Bogdan-Andrei,</pre>
<pre class="yiv7416488403ydp632f2846yiv4421722051moz-signature" style="white-space:pre-wrap;">Thanks for the response! it motivated me to read more stuffs related to opensips. I followed the steps of installation steps and type of route define in <a rel="nofollow" shape="rect" target="_blank" href="https://www.opensips.org/Documentation/Configure-File-2-2" moz-do-not-send="true">https://www.opensips.org/</a> </pre>
<pre class="yiv7416488403ydp632f2846yiv4421722051moz-signature" style="white-space:pre-wrap;">but again i am facing problem in my first installation of stepup </pre>
<pre class="yiv7416488403ydp632f2846yiv4421722051moz-signature" style="white-space:pre-wrap;">Earlier the opensip server was behind the NAT - Public address on firewall and opensip machine (Centos) having a local address (192.168.2.x) , but now the opensips machine (version 2.2) directly hosting public address without mapping ( Lan cable from fortigate firewall to opensips machine with any mapping or port block all ports open)</pre>
<pre class="yiv7416488403ydp632f2846yiv4421722051moz-signature" style="white-space:pre-wrap;">Scenario is </pre>
<pre class="yiv7416488403ydp632f2846yiv4421722051moz-signature" style="white-space:pre-wrap;">If my UAC1 (zoiper on my laptop1) , UAC2 (zoiper on my laptop2 ) and Opensip machine are in the same network i.e ISP1 i can easily hear the voice / audio between the UAC1 and UAC2</pre>
<pre class="yiv7416488403ydp632f2846yiv4421722051moz-signature" style="white-space:pre-wrap;">#) But i change the Network of UAC1 (zoiper on my laptop1) , UAC2 (zoiper on my laptop2 ) to connect with ISP2 and opensip machine remain on ISP1 then i am able to register and call UAC but their is no voice / audio ( RTP Media) among the user agent. ( i am 2 ISP network from different provider) </pre>
<pre class="yiv7416488403ydp632f2846yiv4421722051moz-signature" style="white-space:pre-wrap;">
</pre>
<pre class="yiv7416488403ydp632f2846yiv4421722051moz-signature" style="white-space:pre-wrap;">I the last 1 week i tried all the solution what ever i am able to find online but still its does work. could you please suggest how to troubleshoot further. additionally is their any repository where i can study more about the route how to they work and how to change / Set the the value in header field of the request / response.</pre>
<pre class="yiv7416488403ydp632f2846yiv4421722051moz-signature" style="white-space:pre-wrap;">guidance and direction at stage will help me move further</pre>
<pre class="yiv7416488403ydp632f2846yiv4421722051moz-signature" style="white-space:pre-wrap;">
</pre>
<pre class="yiv7416488403ydp632f2846yiv4421722051moz-signature" style="white-space:pre-wrap;">Thanks in advance , i know few are my quires are wired as i just enter in the world of opensips</pre>
<pre class="yiv7416488403ydp632f2846yiv4421722051moz-signature" style="white-space:pre-wrap;">
</pre>
<pre class="yiv7416488403ydp632f2846yiv4421722051moz-signature" style="white-space:pre-wrap;">
</pre>
<pre class="yiv7416488403ydp632f2846yiv4421722051moz-signature" style="white-space:pre-wrap;">Thanks</pre>
<pre class="yiv7416488403ydp632f2846yiv4421722051moz-signature" style="white-space:pre-wrap;">Sanjeev Kr Sharma</pre>
<pre class="yiv7416488403ydp632f2846yiv4421722051moz-signature" style="white-space:pre-wrap;">
</pre>
<pre class="yiv7416488403ydp632f2846yiv4421722051moz-signature" style="white-space:pre-wrap;">
</pre>
<pre class="yiv7416488403ydp632f2846yiv4421722051moz-signature" style="white-space:pre-wrap;">
</pre>
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<div class="yiv7416488403yahoo_quoted"
id="yiv7416488403yahoo_quoted_3372471652">
<div style="font-family:'Helvetica Neue',
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<div> On Tuesday, 24 July, 2018, 8:11:16 PM
GMT+10, Bogdan-Andrei Iancu
<a class="moz-txt-link-rfc2396E" href="mailto:bogdan@opensips.org"><bogdan@opensips.org></a> wrote: </div>
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<div> <tt>Hi Sanjeev,<br clear="none">
<br clear="none">
As I understand correctly, you end up
connecting into your opensips devices
from different networks - a devices
from the private network (same as
opensips) and another device from the
public network.</tt><br clear="none">
<tt><tt> But note that bidirectional
direct communication between the 2
devices is not possible, as the
public device cannot send traffic to
a private IP/destination.<br
clear="none">
</tt>Depending on the opensips cfg,
the SIP signaling works, as OpenSIPS
will act as a bridge between the 2
networks. But this is not true for
RTP, as RTP goes directly between the
2 devices. So, what you need is to use
a media relay acting a bridge between
the 2 networks.<br clear="none">
<br clear="none">
Regards,<br clear="none">
</tt>
<pre class="yiv7416488403moz-signature">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a rel="nofollow" shape="rect" class="yiv7416488403moz-txt-link-freetext" target="_blank" href="http://www.opensips-solutions.com" moz-do-not-send="true">http://www.opensips-solutions.com</a>
OpenSIPS Bootcamp 2018
<a rel="nofollow" shape="rect" class="yiv7416488403moz-txt-link-freetext" target="_blank" href="http://opensips.org/training/OpenSIPS_Bootcamp_2018/" moz-do-not-send="true">http://opensips.org/training/OpenSIPS_Bootcamp_2018/</a>
</pre>
<div class="yiv7416488403yqt6317164681"
id="yiv7416488403yqt42287">
<div
class="yiv7416488403moz-cite-prefix">On
07/24/2018 05:51 AM, Sanjeev Sharma
via Users wrote:<br clear="none">
</div>
<blockquote type="cite">
<div style="font-family:verdana,
helvetica,
sans-serif;font-size:13px;">
<div>Hi ,</div>
<div><br clear="none">
</div>
<div>i am new to opensips world
and i installed the openSIPS
version 2.2 , facing issue of
media. Setup is my configuration
is like </div>
<div>1) UAC request come through
fortigate firewall on public
address (<a rel="nofollow"
shape="rect"
class="yiv7416488403moz-txt-link-abbreviated"
ymailto="mailto:1234@49.121.121.121" target="_blank"
href="mailto:1234@49.121.121.121"
moz-do-not-send="true">1234@49.121.121.121</a>)
and then passed to my opensip
machine (centos 7 , opensip
version 2.2) having local
address ( natted with public
address) </div>
<div>2) registration of UAC is
fine and if UAC are on same
network lets say one client on
my laptop and another on mobile
device (both on same ISP WIFI)
then voice is going through
between the UAC but i switch the
network of the mobile to telco
service provider then their is
no media pass on in between the
UA or some time only one UA able
to listen the other side.</div>
<div><br clear="none">
</div>
<div>i looked online and search
lots of stuffs related to this
and changed in configuration but
unable to solve or find what and
where i am getting. i tried TCP
dump for both kinds of call i.e
having voice or no voice but
unable to identify the
difference between calls having
voice and no voice.</div>
<div><br clear="none">
</div>
<div>Since i am new to this setup
and configuration , just stuck
due to above problem .</div>
<div><br clear="none">
</div>
<div>Please suggest where to look
and what could be the possible
reason. Currently all traffic (
i.e all ports open) being allow
from firewall to machine </div>
<div><br clear="none">
</div>
<div><br clear="none">
</div>
<div>Thanks</div>
<div>Sanjeev</div>
<div> </div>
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<pre>_______________________________________________
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