<div dir="auto">You need to also</div><div dir="auto"><br></div><div dir="auto">INVITE</div><div dir="auto"><br></div><div dir="auto">Asterisk->kamailio offer the asterisk’s public IP in the SDP.</div><div dir="auto">Kamailio->telco offer in the invite the rtpproxy’s prívate IP (10.x.x.x) </div><div dir="auto"><br></div><div dir="auto">And on the 200</div><div dir="auto"><br></div><div dir="auto">Kamailio->asterisk offer the rtpproxy’s PUBLIC ip.</div><div dir="auto"><br></div><div dir="auto">On the cloud firewall, you need to allow the rtpproxy and Kamailio’s public IPs.</div><div dir="auto">On the asterisk, you need to configure its public IP as the SDP, not the private.</div><div dir="auto"><br></div><div dir="auto">You also need to start rtpproxy in bridge mode providing both the public and private IPs at start (or config)</div><div dir="auto"><br></div><div dir="auto">Hope this helps.</div><div dir="auto"><br><div class="gmail_quote" dir="auto"><div dir="ltr" class="gmail_attr">On Tue, 11 May 2021 at 09:32, Daniel Donoghue <<a href="mailto:daniel.donoghue@freespee.com">daniel.donoghue@freespee.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word;line-break:after-white-space">Since your asterisk server is in the cloud, make sure the relevant udp audio port range is open to the telecoms carrier.<div><br></div><div>It is possible that your firewall/security group setup allows an incoming reply on a port when something first goes out on that port which can explain why audio works in one direction but not the other.</div><div><br></div><div>Blessings,</div><div>—</div></div><div style="word-wrap:break-word;line-break:after-white-space"><div>Daniel</div><div><br></div><div><br><div><div><br><blockquote type="cite"><div>On 11 May 2021, at 10:14, Kashish Raheja <<a href="mailto:kashishraheja1809@gmail.com" target="_blank">kashishraheja1809@gmail.com</a>> wrote:</div><br><div><div dir="ltr"><div dir="ltr"><div dir="ltr"><span style="color:rgb(80,0,80)">This is how it looks:<br><br><ul><li style="margin-left:15px">Asterisk is running on Cloud having a public IP (3.236.X.X)</li><li style="margin-left:15px">Kamailio is running on a Physical server having 2 NIC ports.</li><ul><li style="margin-left:15px">One of them is connected to the SIP trunk and this NIC port local IP is 10.0.87.X. We register to the SBC server (10.0.76.X) of telecom operator carrier through this port.</li><li style="margin-left:15px">Second NIC port is connected to ILL for internet connection having local IP as 192.168.0.192 and public IP as 14.X.X.X</li></ul><li style="margin-left:15px">To make an outbound call, Asterisk Server (3.236.X.X) sends the call to Kamailio server on public IP (14.X.X.X) and in turn Kamailio server sends the call to telecom operator SBC (10.0.76.X) through 10.0.87.X port.</li></ul></span><div>Here is the diagram:</div><div><br></div></div><div dir="ltr"><div><div style="margin:5px 0px"><div id="m_7998670634890728933gmail-q_1619" style="background-color:rgb(232,234,237);border:none;clear:both;line-height:6px;outline:none;width:24px;color:rgb(80,0,80);font-size:11px;border-radius:5.5px"><div style="background:url("https://www.gstatic.com/images/icons/material/system/2x/more_horiz_black_20dp.png") 50% 50%/20px no-repeat;height:11px;opacity:0.7;width:24px"></div></div></div><div style="color:rgb(80,0,80)"><pre style="white-space:pre-wrap;color:rgb(51,51,51);margin-top:0px;margin-bottom:0px;line-height:16.25px"> <a href="http://3.236.72.101:5060/" target="_blank">3.236.72.101:5060</a> <a href="http://192.168.0.192:5060/" target="_blank">192.168.0.192:5060</a> <a href="http://10.0.87.230:5060/" target="_blank">10.0.87.230:5060</a> <a href="http://10.0.76.9:5060/" target="_blank">10.0.76.9:5060</a>
──────────┬───────── ──────────┬───────── ──────────┬───────── ──────────┬─────────
20:24:11.644416 │ INVITE (SDP) │ │ │
+0.000585 │ ──────────────────────────> │ │ │
20:24:11.645001 │ 100 trying -- your call is │ │ │
+0.000235 │ <────────────────────────── │ │ │
20:24:11.645236 │ │ │ INVITE (SDP) │
+0.005768 │ │ │ ──────────────────────────> │
20:24:11.651004 │ │ │ 100 Trying │
+0.580627 │ │ │ <────────────────────────── │
20:24:12.231631 │ │ │ 183 Session Progress (SDP) │
+0.000159 │ │ │ <────────────────────────── │
20:24:12.231790 │ 183 Session Progress (SDP) │ │ │
+1.932655 │ <────────────────────────── │ │ │
20:24:14.164445 │ │ │ 180 Ringing │
+0.000204 │ │ │ <────────────────────────── │
20:24:14.164649 │ 180 Ringing │ │ │
+3.631157 │ <────────────────────────── │ │ │
20:24:17.795806 │ │ │ 200 OK (SDP) │
+0.000361 │ │ │ <────────────────────────── │
20:24:17.796167 │ 200 OK (SDP) │ │ │
+0.233102 │ <────────────────────────── │ │ │
20:24:18.029269 │ ACK │ │ │
+0.000385 │ ──────────────────────────> │ │ │
20:24:18.029654 │ │ │ ACK │
+11.647190 │ │ │ ──────────────────────────> │
20:24:29.676844 │ │ │ BYE │
+0.000605 │ │ │ <────────────────────────── │
20:24:29.677449 │ BYE │ │ │
+0.236993 │ <────────────────────────── │ │ │
20:24:29.914442 │ 200 OK │ │ │
+0.000225 │ ──────────────────────────> │ │ │
20:24:29.914667 │ │ │ 200 OK │
│ │ │ ──────────────────────────> │ </pre></div></div><div><div dir="ltr"><div dir="ltr"><div><div dir="ltr"><div><br></div><div>Thanks.</div><div>Regards</div><div>Kashish</div></div></div></div></div></div></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Mon, May 10, 2021 at 8:26 PM Kashish Raheja <<a href="mailto:kashishraheja1809@gmail.com" target="_blank">kashishraheja1809@gmail.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"><div dir="ltr">Yes, the telecom operator is on the private network. 10.0.X.X is the SBC IP of the telecom operator to which we register. 10.0.X.X is reachable only through the second network interface. The complete flow is given below:<div><br></div><div></div><div><br></div><div>Here 3.236.72.101 is Asterisk Server, 192.168.0.192 is local first network interface IP, 10.0.87.230 is second network interface IP and 10.0.76.9 is telecom operator SBC IP to which we do SIP register.</div><div><br></div><div>We are running the rtpproxy on local IP (192.168.0.192) in the following way:</div><div><br></div><div><b>rtpproxy -F -p /var/run/rtpproxy.pid -u asterisk -l 192.168.0.192 -s udp:localhost:7722</b><br></div><div><br></div><div><br></div><div>Thanks.<br><div><div dir="ltr"><div dir="ltr"><div dir="ltr"><div>Regards</div><div>Kashish<br></div></div></div></div></div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Mon, May 10, 2021 at 4:04 PM Kashish Raheja <<a href="mailto:kashishraheja1809@gmail.com" target="_blank">kashishraheja1809@gmail.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"><div dir="ltr">Here are the SIP Traces:<div><br></div><div><b>Asterisk Server to Kamailio Server (SDP Packet):</b></div><div><br></div><div>2021/05/10 15:54:52.835255 10.0.X.X:5060 -> 10.0.X.X:5060<br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 192.168.0.192;branch=z9hG4bK2599.de1bcd2ba5f8bfc86afb083b0a9e3f65.0;received=10.0.X.X;rport=5060,SIP/2.0/UDP 3.236.X.X:5060;branch=z9hG4bK5a69547a;received=3.236.X.X;rport=5060<br>Record-Route: <<a>sip:192.168.0.192;lr;ftag=as2b21d944</a>><br>Call-ID: <a href="mailto:58eb00885daef7ff3a67ad0e235e817a@14.98.22.110" target="_blank">58eb00885daef7ff3a67ad0e235e817a@14.98.22.110</a><br>From: <<a>sip:68XXXXX@10.0.X.X</a>>;tag=as2b21d944<br>To: <<a href="http://sip:09413745250@192.168.0.192:5060/" target="_blank">sip:09413745250@192.168.0.192:5060</a>>;tag=aa2c806-Huku2c07186a1<br>CSeq: 102 INVITE<br>Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE<br>Contact: <<a>sip:09413745250@10.0.X.X:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff</a>><br>User-Agent: ZTE Softswitch/1.0.0<br>Require: timer<br>Session-Expires: 7200;refresher=uac<br>Content-Length: 182<br>Content-Type: application/sdp<br><br>v=0<br>o=- 1936 20890 IN IP4 10.0.X.X<br>s=SBC call<br>c=IN IP4 10.0.X.X<br>t=0 0<br>m=audio 37874 RTP/AVP 8 101<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=rtpmap:8 PCMA/8000/1<br></div><div><br></div><div><br></div><div><b>Kamailio Server to Telecom Operator Carrier (SDP Packet):</b></div><div><br></div><div>2021/05/10 15:54:52.835419 <a href="http://192.168.0.192:5060/" target="_blank">192.168.0.192:5060</a> -> 3.X.X.X:5060<br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 3.236.72.101:5060;branch=z9hG4bK5a69547a;received=3.236.72.101;rport=5060<br>Record-Route: <<a>sip:192.168.0.192;lr;ftag=as2b21d944</a>><br>Call-ID: <a href="mailto:58eb00885daef7ff3a67ad0e235e817a@14.98.22.110" target="_blank">58eb00885daef7ff3a67ad0e235e817a@14.98.22.110</a><br>From: <<a>sip:68XXXXX@10.0.X.X</a>>;tag=as2b21d944<br>To: <<a href="http://sip:09413745250@192.168.0.192:5060/" target="_blank">sip:09413745250@192.168.0.192:5060</a>>;tag=aa2c806-Huku2c07186a1<br>CSeq: 102 INVITE<br>Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE<br>Contact: <<a>sip:09413745250@10.0.X.X:5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff</a>><br>User-Agent: ZTE Softswitch/1.0.0<br>Require: timer<br>Session-Expires: 7200;refresher=uac<br>Content-Length: 182<br>Content-Type: application/sdp<br><br>v=0<br>o=- 1936 20890 IN IP4 10.0.X.X<br>s=SBC call<br>c=IN IP4 10.0.X.X<br>t=0 0<br>m=audio 37874 RTP/AVP 8 101<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=rtpmap:8 PCMA/8000/1<br></div><div><br></div><div><div><div dir="ltr"><div dir="ltr"><div dir="ltr"><div>Regards</div><div>Kashish</div></div></div></div></div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Mon, May 10, 2021 at 2:37 PM Kashish Raheja <<a href="mailto:kashishraheja1809@gmail.com" target="_blank">kashishraheja1809@gmail.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"><div dir="ltr">Hi All,<div><br></div><div>I have set up Kamailio in the following manner:</div><div><br></div><div>Kamailio (Physical Server: Register to Telecom Operator Carrier SIP trunk) ---> Asterisk Server (on Cloud having public IP)</div><div><br></div><div>I am successfully able to route the call to Asterisk server on Cloud when I make a call to the number provided by the carrier and there is audio also on both sides.</div><div><br></div><div>However, when I am making an outbound call from Asterisk server to the number through Kamailio, there is no audio when I pick up the call. I have tried to capture the traces but not able to understand the exact problem here.</div><div><br></div><div>Note: I am running the RTP proxy on Kamailio server.</div><div><br></div><div>Any help on why this might be happening?</div><div><br></div><div>Thanks.<br clear="all"><div><div dir="ltr"><div dir="ltr"><div><div dir="ltr"><div>Regards</div><div>Kashish</div><div>+919413745250</div></div></div></div></div></div></div></div>
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</blockquote></div></div>-- <br><div dir="ltr" class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div>Regards,</div><div><br></div>David Villasmil<div>email: <a href="mailto:david.villasmil.work@gmail.com" target="_blank">david.villasmil.work@gmail.com</a></div><div>phone: +34669448337</div></div></div>