Hi<div><br /></div><div>- I see you mentioned rtpengine but config shows rtpproxy - which one you use? Maybe you mixed things?</div><div>- Have you tried to capture internal interface of kamailio machine? Is RTP there?<br /><br />-- <br />Wbr, Serge via mobile<br /><br />09.01.2019, 10:37, "Prashant Gupta" &lt;prashant@farmguide.in&gt;:<br /><blockquote type="cite"><div dir="ltr">Hi,<div>I have the following architecture - SIP provider &lt;-&gt; Kamailio &lt;-&gt; Asterisk servers<br /></div><div>Currently I have everything setup and incoming calls from Sip are routed to my asterisk server. The issue is however that when I answer the call, there is no media in the call. I have tried connecting with a normal local extension(not SIP,eg 1001) and there is a normal flow of media.</div><div>When i try to sniff my connection via Wireshark on the asterisk server, there is an outflow of RTP packets but the same RTP traffic does not appear on the Wireshark of my Kamailio server connection. </div><div>I am not sure if this is an RTP engine issue and how to resolve this.</div><div>I have -</div><div>modparam("rtpproxy", "rtpproxy_sock", "udp:<a href="http://127.0.0.1:45038/" target="_blank">127.0.0.1:45038</a>")<br /></div><div>this in my kamailio cfg but I don;t know which port to use here.</div><div>Any suggestions?</div></div>
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