<html>
<head>
<style><!--
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 10pt;
font-family:Tahoma
}
--></style></head>
<body class='hmmessage'><div dir='ltr'>
Thank you Steve - here it is - http://pastebin.freeswitch.org/19510.<br><br>Fresh call, new ports.<br><br>Notice this line, this is what I want to stop:<br>2012-07-13 10:18:42.935692 [INFO] switch_rtp.c:3133 Auto Changing port from 10.3.212.105:8024 to 10.3.212.105:12016<br><br>Thank you,<br>A.C.<br><br><div><div id="SkyDrivePlaceholder"></div><hr id="stopSpelling">CC: freeswitch-users@lists.freeswitch.org<br>From: steveayre@gmail.com<br>Date: Fri, 13 Jul 2012 13:30:38 +0100<br>To: freeswitch-users@lists.freeswitch.org<br>Subject: Re: [Freeswitch-users] One Way RTP (no NAT) - Freeswitch sends RTP to the remote source port instead of SDP negotiated destination port (with disable_rtp_auto_adjust=true)<br><br><div>Can you supply a pastebin of adebug log of the call?</div><div><br>Steve on iPhone</div><div><br></div><div><br>On 12 Jul 2012, at 21:39, Adelia C. <<a href="mailto:hexade@hotmail.com">hexade@hotmail.com</a>> wrote:<br><br></div><div></div><blockquote><div>
<style><!--
.ExternalClass .ecxhmmessage P
{padding:0px;}
.ExternalClass body.ecxhmmessage
{font-size:10pt;font-family:Tahoma;}
--></style>
<div dir="ltr">
<font style="font-size:12pt" face="Garamond" size="3">We're testing a feature that requires an extra Application Server (B2BUA) on our call path, on the egress side of the call from FreeSwitch. </font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3">With the extra AS, we have only one-way RTP (Caller -> Callee). Without, RTP is fine. </font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3">We narrowed it down to FS enforcing symmetric RTP while AS only works with asymmetric RTP. We cannot change the behavior of the AS at this time.</font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3">Call: </font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3">Caller - >NTW -> SBC -> FS -> AS -> SBC -> NTW -> Callee.</font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3">RTP streams, as negotiated (SDP):</font><font style="font-size:12pt" face="Garamond" size="3"><br></font>
<p class="ecxMsoNormal" style="margin-left:0.5in;text-indent:0.5in"><font style="font-size:12pt" face="Garamond" size="3"><span style="font-size:14pt;font-family:"Garamond","serif""><br></span></font></p><p class="ecxMsoNormal" style="margin-left:.5in;text-indent:.5in"><font style="font-size:12pt" face="Garamond" size="3"><span style="font-size:14.0pt;font-family:"Garamond","serif"">Caller - > SBCOut:50460
->FSIn:29526 -> <b>FSOut:17818 -> ASIn:8080</b> -> ASOutB:19040 -> SBCOut:49162 -> Callee (can't hear a thing)<br></span></font></p>
<p class="ecxMsoNormal" style="margin-left:.5in;text-indent:.5in"><font style="font-size:12pt" face="Garamond" size="3"><span style="font-size:14.0pt;font-family:"Garamond","serif"">Callee -> SBCOut:49162
->ASIn:8082 -> <b>ASOut:19000 -> FSIn:17818</b> -> FSOut:29526 -> SBCIn:50460 -> Caller (good RTP)<br></span></font></p>
<p class="ecxMsoNormal"><font style="font-size:12pt" face="Garamond" size="3"><span style="font-size:14.0pt;font-family:"Garamond","serif""> </span></font></p><font style="font-size:12pt" face="Garamond" size="3">RTP streams, as sent/receiced:<br><br>FS negotiates 8080 as the RTP destination <b>to</b> AS in the "caller - > callee" direction. For about .5s, that's how the RTP flows : </font><font style="font-size:12pt" face="Garamond" size="3"><span style="font-size:14.0pt;font-family:"Garamond","serif""><b>FSOut:17818 -> ASIn:8080</b></span></font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3">FS negotiates 19000 as the RTP source <b>from </b>AS in the "callee -> caller" direction. This is how the RTP flows for the duration of the call.: </font><font style="font-size:12pt" face="Garamond" size="3"><span style="font-size:14pt;font-family:"Garamond","serif""><b>ASOut:19000 -> FSIn:17818<br></b></span></font><font style="font-size:12pt" face="Garamond" size="3">FS switches its RTP stream in the "caller - > callee" direction ~.5s into the call : </font><font style="font-size:12pt" face="Garamond" size="3"><span style="font-size:14pt;font-family:"Garamond","serif""><b>FSOut:17818 -> ASIn:</b></span></font><font style="font-size:12pt" face="Garamond" size="3"><span style="font-size:14pt;font-family:"Garamond","serif""><b>19000. [</b></span></font><font style="font-size:12pt" face="Garamond" size="3">ASIn listens on 8080, the caller - > called RTP is dead.</font><font style="font-size:12pt" face="Garamond" size="3">]<br></font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3">All this from Wireshark trace.</font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3">I tried the following, with no success:</font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3">1. Added to MyFSApp.xml and dialplan\public.xml:</font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3"> <action application="set" data="disable_rtp_auto_adjust=true"/></font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3"> <action application="set" data="rtp_manual_rtp_bugs=accept_any_packets"/> </font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3">Restarted, FS, same result.</font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3">2. Commented out </font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3"><param name="ext-rtp-ip" value="auto-nat"/></font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3"><param name="ext-sip-ip" value="auto-nat"/></font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3"><param name="use-rtp-timer" value="true"/></font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3">Restarted, FS, same result.</font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3">3. Started FS with -nonat, then nonat</font><font style="font-size:12pt" face="Garamond" size="3">map</font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3">Same result.</font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3">Other info : </font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3">- </font><font style="font-size:12pt" face="Garamond" size="3">Our App on FS runs in B2BUA mode, doesn't proxy the call. We answers the incoming leg before initiating the outgoing leg.</font><font style="font-size:12pt" face="Garamond" size="3"> </font><font style="font-size:12pt" face="Garamond" size="3">I can attach the Wireshark is I didn't provide enough info - I have both FS and AS pcaps.</font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3">- Our FS code about 2 weeks old.</font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3">What else can I try? Any other settings I can enable/disable? Can I enable extra RTP logs on FreeSwitch for debug? </font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3">Thanks,</font><font style="font-size:12pt" face="Garamond" size="3"><br></font><font style="font-size:12pt" face="Garamond" size="3">A.C.</font><br> </div>
</div></blockquote><blockquote><div><span>_________________________________________________________________________</span><br><span>Professional FreeSWITCH Consulting Services:</span><br><span><a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a></span><br><span><a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a></span><br><span></span><br><span>FreeSWITCH-powered IP PBX: The CudaTel Communication Server</span><br><span><a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a></span><br><span></span><br><span>Official FreeSWITCH Sites</span><br><span><a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a></span><br><span><a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a></span><br><span><a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a></span><br><span></span><br><span>Join Us At ClueCon - Aug 7-9, 2012</span><br><span></span><br><span>FreeSWITCH-users mailing list</span><br><span><a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a></span><br><span><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a></span><br><span>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users</span><br><span><a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a></span><br></div></blockquote><br>_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org</div> </div></body>
</html>