<div><div>Corrigendum:</div></div><div dir="auto">We are getting packets from CALLER*<br></div><div dir="auto"><br></div><div dir="auto">Regards</div><div dir="auto">Abbasi</div><div><br><div class="gmail_quote"><div>On Fri, 18 May 2018 at 12:46 PM, Bilal Abbasi <<a href="mailto:bilaln018@gmail.com">bilaln018@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="auto">Actually they dont even get out of rtpproxy, we cant see that in the in sngrep/tcpdump.<div dir="auto">We are getting packets from callee but nothing going out of rtpproxy(talking about local dump)</div><div dir="auto"><br></div><div dir="auto">Regards</div></div><div dir="auto"><div dir="auto">Abbasi</div></div><br><div class="gmail_quote"><div>On Fri, May 18, 2018, 12:13 Giovanni Tommasini - evosip <giovanni.tommasini@evosip.cloud> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="gmail_default" style="font-family:"trebuchet ms",sans-serif">Hi Younas,</div><div class="gmail_default" style="font-family:"trebuchet ms",sans-serif"><br></div><div class="gmail_default"><div class="gmail_default"><span style="font-family:"trebuchet ms",sans-serif">when you have a call if you make a trace <span style="color:rgb(34,34,34);font-family:"trebuchet ms",sans-serif;font-size:small;font-style:normal;font-variant-ligatures:normal;font-variant-caps:normal;font-weight:400;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;background-color:rgb(255,255,255);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline">in your sip server<span> </span></span>with sngrep or tshark can you see the RTP packet ?</span><br></div><div class="gmail_default"><span style="font-family:"trebuchet ms",sans-serif">you says "<span style="color:rgb(34,34,34);font-family:arial,sans-serif;font-size:12.8px;font-style:normal;font-variant-ligatures:normal;font-variant-caps:normal;font-weight:400;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px;background-color:rgb(255,255,255);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline">I have kamailio server behind nat with rptproxy.", so is it possible that the router in front your SIP server blocks the traffic? </span></span><span style="font-size:12.8px">could you have a trace there?</span></div><div class="gmail_default"><span style="font-size:12.8px">I mean </span><span style="font-size:12.8px">the packets exit the rtpproxy but don't arrive to callee or just don't come out of the rtpproxy?</span></div></div><div class="gmail_extra"><br clear="all"><div><div class="m_-3640671945666690073m_2629072283700475430gmail_signature" data-smartmail="gmail_signature"><div><pre style="color:rgb(0,0,0);font-family:"Lucida Grande",Verdana,Arial,Helvetica,sans-serif;font-size:12px;margin-top:0px;margin-bottom:0px;padding:0px;word-wrap:break-word;line-height:normal;white-space:pre-wrap!important"><span style="font-family:"courier new",courier"><span>Giovanni</span> <span>Tommasini</span> | <a href="http://evosip.cloud" rel="noreferrer" target="_blank"><b><div class="gmail_default" style="font-family:"trebuchet ms",sans-serif;display:inline"></div>evosip</b></a></span></pre></div></div></div>
<br><div class="gmail_quote">2018-05-18 1:36 GMT+02:00 Aqs Younas <span><<a href="mailto:aqsyounas@gmail.com" rel="noreferrer" target="_blank">aqsyounas@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div dir="auto">Thanks for replying. Callee is actually a sip provider. I see no firewall or iptables rules on server to prevent rtp following towards callee.</div><div dir="auto"><br></div><div dir="auto">Is there anything i can do to see which thing is blocking rtps.</div><div dir="auto"><br></div><div dir="auto">Best</div><div><div class="m_-3640671945666690073m_2629072283700475430h5"><br><div class="gmail_quote"><div>On Fri, 18 May 2018 at 3:59 AM, Mack Hendricks <<a href="mailto:ap@goflyball.com" rel="noreferrer" target="_blank">ap@goflyball.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"><div>Is there something blocking RTP traffic from reaching the callee? Is the callee a carrier or the actually endpoint endpoint (aka SIP Phone)?</div><br><div>
<div dir="auto" style="word-wrap:break-word;line-break:after-white-space"><div style="color:rgb(0,0,0);font-family:Helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><b>Mack Hendricks / Head of Support / dOpenSource</b></div><div style="color:rgb(0,0,0);font-family:Helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px">web: <a href="http://dopensource.com" rel="noreferrer" target="_blank">http://dopensource.com</a></div><div style="color:rgb(0,0,0);font-family:Helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px">support: +888-907-2085</div><div style="color:rgb(0,0,0);font-family:Helvetica;font-size:12px;font-style:normal;font-variant-caps:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><a href="http://dsiprouter.org" rel="noreferrer" target="_blank">dSIPRouter</a> - GUI focused on implementing Kamailio to provide SIP Trunking and PBX Hosting Services</div></div>
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<div><br><blockquote type="cite"></blockquote></div></div><div style="word-wrap:break-word;line-break:after-white-space"><div><blockquote type="cite"><div>On May 18, 2018, at 12:50 AM, Aqs Younas <<a href="mailto:aqsyounas@gmail.com" rel="noreferrer" target="_blank">aqsyounas@gmail.com</a>> wrote:</div><br class="m_-3640671945666690073m_2629072283700475430m_-5510585572842309696m_-6090413338680338134Apple-interchange-newline"></blockquote></div></div><div style="word-wrap:break-word;line-break:after-white-space"><div><blockquote type="cite"><div><div>Greetings list, <div><br></div><div><br></div><div>I have kamailio server behind nat with rptproxy. But i am getting no voice on the call. After taking trace i could see that rtpproxy was getting rtp packets but not packets was being forwarded towards callee side. </div><div><br></div><div>Though i see in rtpproxy logs, packets being relayed from caller side. </div><div><br></div><div>Does rtpproxy wait to receive a single rtp frame from callee before fowarding that to callee?<br></div><div><br></div><div>Since, i am getting no rtp from callee. So, might be rptrpoxy is waiting to learn source address of callee side rtp, that is why it is not forwarding rtp packets from caller towards callee. </div><div><br></div><div><br></div><div>Any pointer/suggestion is much appreciated.</div><div><br></div><div>Best Regards, </div><div><br></div><div>Aqs Younas</div></div></div></blockquote></div></div><div style="word-wrap:break-word;line-break:after-white-space"><div><blockquote type="cite"><div>
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