<html><head></head><body><div style="font-family:verdana, helvetica, sans-serif;font-size:13px;"><div style="font-family:verdana, helvetica, sans-serif;font-size:13px;"><div></div>
<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 sip:124007@149.355.453.253;transport=UDP 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: <sip:124010@159.200.37.234:5060;transport=UDP>.</div><div>To: <sip:124007@149.355.453.253;transport=UDP>.</div><div>From: <sip:124010@149.355.453.253;transport=UDP>;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 <sanjeevt510@yahoo.com> wrote:
</div>
<div><br></div>
<div><br></div>
<div><div id="yiv7416488403"><div><div style="font-family:verdana, helvetica, sans-serif;font-size:13px;"><div style="font-family:verdana, helvetica, sans-serif;font-size:13px;"><div></div>
<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">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;"><br clear="none"></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;"><br clear="none"></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;"><br clear="none"></pre><pre class="yiv7416488403ydp632f2846yiv4421722051moz-signature" style="white-space:pre-wrap;"><br clear="none"></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;"><br clear="none"></pre><pre class="yiv7416488403ydp632f2846yiv4421722051moz-signature" style="white-space:pre-wrap;"><br clear="none"></pre><pre class="yiv7416488403ydp632f2846yiv4421722051moz-signature" style="white-space:pre-wrap;"><br clear="none"></pre><div><br clear="none"></div><br clear="none"></div><div><br clear="none"></div>
</div><div class="yiv7416488403yqt3632924350" id="yiv7416488403yqt99726"><div class="yiv7416488403yahoo_quoted" id="yiv7416488403yahoo_quoted_3372471652">
<div style="font-family:'Helvetica Neue', Helvetica, Arial, sans-serif;font-size:13px;color:#26282a;">
<div>
On Tuesday, 24 July, 2018, 8:11:16 PM GMT+10, Bogdan-Andrei Iancu <bogdan@opensips.org> wrote:
</div>
<div><br clear="none"></div>
<div><br clear="none"></div>
<div><div id="yiv7416488403"><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">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/">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">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>
<div><br clear="none">
</div>
<div><br clear="none">
</div>
<div><br clear="none">
</div>
</div>
<br clear="none">
<fieldset class="yiv7416488403mimeAttachmentHeader"></fieldset>
<br clear="none">
<pre>_______________________________________________
Users mailing list
<a rel="nofollow" shape="rect" class="yiv7416488403moz-txt-link-abbreviated" ymailto="mailto:Users@lists.opensips.org" target="_blank" href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a>
<a rel="nofollow" shape="rect" class="yiv7416488403moz-txt-link-freetext" target="_blank" href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a>
</pre>
</blockquote></div>
<br clear="none">
</div></div></div>
</div>
</div></div></div></div></div></div>
</div>
</div></div></body></html>