<div dir="ltr"><div dir="ltr"><div class="gmail_quote"><div dir="ltr" class="gmail_attr">El jue., 28 de feb. de 2019 a la(s) 09:21, Daniel Tryba (<a href="mailto:d.tryba@pocos.nl">d.tryba@pocos.nl</a>) escribió:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">You wrote something else in your original message:<br>
exposing asterisk RTP 10000-30000 makes it function. <br></blockquote><div>that's the problem here! yes!<br></div><div> </div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
I read that to be an indication that rtp(proxy|engine) isn't rewriting.<br></blockquote><div>in fact, the first try was with rtpproxy, but due the DEBUG log are very poor<br></div><div>we tried with rtpengine, and now i noted that rtpengine only output in log that:<br>```<br><span style="font-family:monospace,monospace">Feb 28 14:39:44 ip-10-10-1-1 rtpengine[28721]: DEBUG: timer run time = 0.000005 sec<br>Feb 28 14:39:44 ip-10-10-1-1 rtpengine[28803]: [1551364784.000249] DEBUG: timer run time = 0.000035 sec<br>Feb 28 14:39:45 ip-10-10-1-1 rtpengine[28864]: DEBUG: timer run time = 0.000010 sec<br>Feb 28 14:39:46 ip-10-10-1-1 rtpengine[28864]: DEBUG: timer run time = 0.000007 sec<br>Feb 28 14:39:46 ip-10-10-1-1 rtpengine[28732]: DEBUG: timer run time = 0.000008 sec<br>Feb 28 14:39:46 ip-10-10-1-1 rtpengine[28832]: DEBUG: timer run time = 0.000005 sec</span><br>```<br><br></div><div>BUT: rtpproxy seems are working cos if not there's no SIP sesion stablished!<br></div><div>but as i siad, no sound if i not exposed the asterisk ports!<br><br></div><div>NOTE: audio of the call are udp! so that means are not possible? i guess must be possible<br></div><div>due the kamailio can manage an route using the rtp[proxy|engine] right?<br></div><div> </div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
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