<html><head><meta http-equiv="Content-Type" content="text/html; charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""><div class="">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 class=""><div class="">
<div dir="auto" style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class=""><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; -webkit-text-stroke-width: 0px;"><b class="">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; -webkit-text-stroke-width: 0px;">web: <a href="http://dopensource.com" class="">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; -webkit-text-stroke-width: 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; -webkit-text-stroke-width: 0px;"><a href="http://dsiprouter.org" class="">dSIPRouter</a> - GUI focused on implementing Kamailio to provide SIP Trunking and PBX Hosting Services</div></div>
</div>
<div><br class=""><blockquote type="cite" class=""><div class="">On May 18, 2018, at 12:50 AM, Aqs Younas <<a href="mailto:aqsyounas@gmail.com" class="">aqsyounas@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div dir="ltr" class="">Greetings list, <div class=""><br class=""></div><div class=""><br class=""></div><div class="">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 class=""><br class=""></div><div class="">Though i see in rtpproxy logs, packets being relayed from caller side. </div><div class=""><br class=""></div><div class="">Does rtpproxy wait to receive a single rtp frame from callee before fowarding that to callee?<br class=""></div><div class=""><br class=""></div><div class="">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 class=""><br class=""></div><div class=""><br class=""></div><div class="">Any pointer/suggestion is much appreciated.</div><div class=""><br class=""></div><div class="">Best Regards, </div><div class=""><br class=""></div><div class="">Aqs Younas</div></div>
_______________________________________________<br class="">Kamailio (SER) - Users Mailing List<br class=""><a href="mailto:sr-users@lists.kamailio.org" class="">sr-users@lists.kamailio.org</a><br class="">https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users<br class=""></div></blockquote></div><br class=""></body></html>