<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Peter,<div><br></div><div>that's not RTP, but RTCP.</div><div>The RTCP port is always RTP+1.</div><div><br><div>
<span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: 14px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: 14px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div><font class="Apple-style-span" face="'Helvetica Neue'"><font class="Apple-style-span" color="#1C00FF">David Ponzone </font><font class="Apple-style-span" color="#000000" size="3"><span class="Apple-style-span" style="font-size: 12px; ">Direction Technique</span></font></font></div><div><font class="Apple-style-span" face="'Helvetica Neue'"><font class="Apple-style-span" size="3"><span class="Apple-style-span" style="font-size: 13px; ">email: <a href="mailto:david.ponzone@ipeva.fr">david.ponzone@ipeva.fr</a></span></font></font></div><div><font class="Apple-style-span" face="'Helvetica Neue'"><font class="Apple-style-span" size="3"><span class="Apple-style-span" style="font-size: 13px; ">tel: 01 74 03 18 97</span></font></font></div><div><font class="Apple-style-span" face="'Helvetica Neue'"><font class="Apple-style-span" size="3"><span class="Apple-style-span" style="font-size: 13px; ">gsm: 06 66 98 76 34</span></font></font></div><div><font class="Apple-style-span" face="'Helvetica Neue'"><br></font></div><div><font class="Apple-style-span" color="#1C00FF" face="'Helvetica Neue'">Service Client<span class="Apple-converted-space"> </span></font><font class="Apple-style-span" face="'Helvetica Neue'"><font class="Apple-style-span" color="#FF0000">IP</font></font><font class="Apple-style-span" color="#1C00FF" face="'Helvetica Neue'">eva</font></div><div><font class="Apple-style-span" color="#1C00FF" face="'Helvetica Neue'"><span class="Apple-style-span" style="color: rgb(0, 0, 0); font-family: Helvetica; "><div><font class="Apple-style-span" face="'Helvetica Neue'"><font class="Apple-style-span" size="3"><span class="Apple-style-span" style="font-size: 13px; ">tel: 0811 46 26 26</span></font></font></div><div><font class="Apple-style-span" face="'Helvetica Neue'" size="3"><span class="Apple-style-span" style="font-size: 13px; "><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; font: normal normal normal 10px/normal Arial; color: rgb(0, 34, 243); "><span style="text-decoration: underline; "><a href="BLOCKED::http://www.ipeva.fr/">www.ipeva.fr</a></span><span style="color: rgb(101, 104, 149); "> - <span style="color: rgb(0, 34, 243); text-decoration: underline; "><a href="BLOCKED::http://www.ipeva-studio.com/">www.ipeva-studio.com</a></span></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; font: normal normal normal 10px/normal Arial; color: rgb(0, 34, 243); "><span class="Apple-style-span" style="text-decoration: underline; "><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; font: normal normal normal 10px/normal Arial; color: rgb(0, 34, 243); "><span class="Apple-style-span"><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; text-align: justify; font: normal normal normal 10px/normal Arial; color: rgb(192, 192, 192); "><i>Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. </i><b><i>IPeva</i></b><i> décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.</i></div><div style="text-decoration: underline; text-align: justify; "><font class="Apple-style-span" color="#C0C0C0"><i><br></i></font></div></span></div></span></font></div></span></font></div></div></span><br class="Apple-interchange-newline"></span><br class="Apple-interchange-newline">
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<br><div><div>Le 14/11/2010 à 22:31, Peter Steinbach a écrit :</div><br class="Apple-interchange-newline"><blockquote type="cite"><div>Hello Yasuro,<br><br>what I am wondering about is: Freeswitch is offering port 16670 in its<br>SDP message of the SIP OK message, and 192.168.11.250 is trying to send<br>RTP packets to 16671 instead. But this Freeswitch port is not open. So<br>192.168.11.250 closes the connection: See<br> 217 49.351287 192.168.11.250 5005 192.168.11.11 <br> 16671 RTCP Receiver Report Source description <br> 218 49.351369 192.168.11.11 5005 192.168.11.250 <br>16671 ICMP Destination unreachable (Port unreachable)<br><br><br><br>-- <br>With kind regards<br>Peter Steinbach <br><br>Telefaks Services GmbH<br>Theo-Geisel-Strasse 25<br>D 61250 Usingen, Germany<br>mailto:lists (att) <a href="http://telefaks.de">telefaks.de</a><br>Internet: <a href="http://www.telefaks.de">www.telefaks.de</a><br><br><br><br>Yasuro schrieb:<br><blockquote type="cite">Michael and other FreeSWITCH gurus:<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">As per your instruction, I have created logs. Please take a look<br></blockquote><blockquote type="cite"><<a href="http://pastebin.freeswitch.org/14479">http://pastebin.freeswitch.org/14479</a>>. It includes FS's messages<br></blockquote><blockquote type="cite">while it initializes itself. The VoIP adapter (SIP gateway, which also<br></blockquote><blockquote type="cite">works as a router)'s IP address is *192.168.11.250* on the LAN side.<br></blockquote><blockquote type="cite">That of the PC which is running FS is *192.168.11.11*. Its firewall<br></blockquote><blockquote type="cite">function was turned off during this testing. The summary of packet<br></blockquote><blockquote type="cite">exchange between the two during this same period is here<br></blockquote><blockquote type="cite"><<a href="http://pastebin.freeswitch.org/14480">http://pastebin.freeswitch.org/14480</a>>. Please see my original posting<br></blockquote><blockquote type="cite">(which is included at the end of this message) for the setup of my<br></blockquote><blockquote type="cite">home LAN.<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">~250 is sending RTCP receiver report before ~11 starts RTP<br></blockquote><blockquote type="cite">communication. I am wondering if the former is giving up on the call<br></blockquote><blockquote type="cite">because of time out; and, if it is true, if there is any way to have<br></blockquote><blockquote type="cite">it wait longer.<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">I am completely stuck about this issue and I'll welcome any input you<br></blockquote><blockquote type="cite">have.<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Thanks!<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Yasuro<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Michael Collins wrote (11/12/2010 4:43 AM):<br></blockquote><blockquote type="cite"><blockquote type="cite">Run your same test on FreeSWITCH but turn on sip debugging at the fs_cli:<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite">sofia global siptrace on<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite">Then make the call and capture the output and put into new pastebin.<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite">Hopefully you'll see why the channel is already hungup when it goes<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite">to play music.<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite">-MC<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite">2010/11/10 Yasuro <<a href="mailto:yasuro@yasuro.com">yasuro@yasuro.com</a> <<a href="mailto:yasuro@yasuro.com">mailto:yasuro@yasuro.com</a>>><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> Hi, FreeSWITCH gurus! I need your help!<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> First off, I am new to FS and I am new to Internet telephony as<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> well. Heck, I am new to the concept of NAT, UPnP, etc., so please<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> bear with my ignorance.<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> I subscribe to a VoIP service at home, with which I get one DID.<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> They supply me a VoIP adapter. Their expected usage is for you to<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> plug in analog phones to the analog phone jacks in the VoIP<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> adapter. However, It also has four Ethernet LAN ports and it acts<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> as a router. You can also access it from the LAN side and<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> register with its built-in SIP gateway.<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> What I would like to do is to run FS (Windows version) on one of<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> the Windows PCs, have it register with the SIP gateway, and have<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> it act as an AA or IVR. For testing, I am having it just play music.<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> When I tried the same idea with AsterikWin32, it worked just as I<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> had hoped; it answered incoming calls automatically. However, I<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> somehow cannot make it work with FS. I simulate incoming calls to<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> my DID number with Skype's Sypeout. It fails after a short while<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> with such error messages as "network error." It appears the call<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> was never answered.<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> FS is assigned an extension number 7 at the gateway. When I call<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> extension 7 from a different extension (at the gateway level, not<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> an extension inside FS), FS does answer the call and I hear<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> music. FS fails to answer only incoming calls from outside.<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> I think my FS configuration is fairly standard. I created an<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> external SIP profile for the gateway under<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> conf/sip_profiles/external/ and modified<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> conf/dialplan/public/00_inbound_did.xml so incoming calls to the<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> gateway will be transferred to an extension within FS.<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> FS's messages and logs, plus the result of packet captures<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> indicate that FS /thinks /it has answered the call, and goes on<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> to initiate media communication. I see RTP packets going from FS<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> to the SIP gateway. What's different from AsteriskWin32's case is<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> that there are no RTP packets coming back from the SIP gateway to<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> FS. Turning of the firewall of the PC does not seem to change the<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> result in any way.<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> For your perusal, I have created the following logs of<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> communication between FS/AsteriskWin32 and the SIP gateway:<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> * AsteriskWin32's case<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> o Summary: <a href="http://pastebin.freeswitch.org/14457">http://pastebin.freeswitch.org/14457</a><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> o Details: <a href="http://pastebin.freeswitch.org/14460">http://pastebin.freeswitch.org/14460</a><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> * FreeSWITCh's case<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> o Summary: <a href="http://pastebin.freeswitch.org/14462">http://pastebin.freeswitch.org/14462</a><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> o Details: <a href="http://pastebin.freeswitch.org/14463">http://pastebin.freeswitch.org/14463</a><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> o Log: <a href="http://pastebin.freeswitch.org/14465">http://pastebin.freeswitch.org/14465</a><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> The IP address of the SIP gateway is *192.168.11.250*, and that<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> of the PC FS/AsteriskWin32 resides in is *192.168.11.11*. My DID<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> number is masked as ABCDEFGHIJ. I do not know if it gives you any<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> useful information, but those files include the registration<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> phase. /FS's log was taken at a different time/, so it does not<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> entirely match the packet captures.<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> I also have the corresponding Pcap files. Please let me know if<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> you need them.<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> I am not entirely sure, but I think as far as what I'd like to do<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> is concerned, NAT is not going to be an issue, because the<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> FS/AsteriskWin32 PC and the SIP gateway (its LAN side IP address)<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> are on the same subnet (192.168.11/24). At this time, I do not<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> need to access FS from the Internet.<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> Finally, I will give you more details about my setup, which may<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> or may not be relevant to this issue.<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> My home LAN is set up this way:<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> <a href="http://i139.photobucket.com/albums/q302/starplatina/For%20Blogs/HomeLANSetup.jpg">http://i139.photobucket.com/albums/q302/starplatina/For%20Blogs/HomeLANSetup.jpg</a><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> Please note that there are /two layers/ of NAT, and that in the<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> inner layer, two NAT devices exist. I know it looks convoluted,<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> but there are logical reasons for this setup.<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> The VoIP service provider only supports the PCMU codec. The music<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> file I prepared for this testing is encoded in PCMU, so codecs<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> will not be an issue.<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> Please do not hesitate to ask if you have any questions. Thanks<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> for your help in advance!<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> Yasuro<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> _______________________________________________<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> FreeSWITCH-users mailing list<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> <<a href="mailto:FreeSWITCH-users@lists.freeswitch.org">mailto:FreeSWITCH-users@lists.freeswitch.org</a>><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"> <a href="http://www.freeswitch.org">http://www.freeswitch.org</a><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite">_______________________________________________<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite">FreeSWITCH-users mailing list<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite">UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><a href="http://www.freeswitch.org">http://www.freeswitch.org</a><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">------------------------------------------------------------------------<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">_______________________________________________<br></blockquote><blockquote type="cite">FreeSWITCH-users mailing list<br></blockquote><blockquote type="cite"><a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br></blockquote><blockquote type="cite"><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br></blockquote><blockquote type="cite">UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br></blockquote><blockquote type="cite"><a href="http://www.freeswitch.org">http://www.freeswitch.org</a><br></blockquote><blockquote type="cite"><br></blockquote><br><br><br>_______________________________________________<br>FreeSWITCH-users mailing list<br><a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>http://www.freeswitch.org<br></div></blockquote></div><br></div></body></html>