<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">We are able to generate cdrs in no media mode and we are able to use sip session timers to detect a dead endpoint. For obvious reasons, you can't do rtp timeout if we are not in the media path.<div><br class="webkit-block-placeholder"></div><div>Mike</div><div><br class="webkit-block-placeholder"></div><div><br><div><div>On Mar 17, 2008, at 8:51 AM, Vijay Ramnarine wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">Would FS be able to do any sort of tracking / billing in No Media Mode ?<br>how about something like rtp timeout?<br><br><div class="gmail_quote">On Sun, Mar 16, 2008 at 11:46 PM, Kurt Marasco <<a href="mailto:kmarasco@faithwork.org">kmarasco@faithwork.org</a>> wrote:<br> <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Thanks for your comments Ken. They were very helpful. This group is<br> great. I always seem to get multiple responses:)<br> <div><div></div><div class="Wj3C7c"><br> Ken Rice wrote:<br> > FreeSwitch has 3 modes of operation when it comes to sip calls. Please Note<br> > FreeSwitch is a B2BUA and not a proxy although it has proxy like modes.<br> ><br> > Mode 1) Full Media Interaction this is what you are seeing. You can<br> > completely interact with the media stream allowing for sniffing the media,<br> > transcoding, or injecting the media for various purposes.<br> ><br> > Mode 2) Media Proxy only mode. In this mode FreeSwitch just acts as a dumb<br> > RTP proxy for things like Topology Hiding and NAT busting. SIP SDPs are<br> > copied across with updates to where to send the media and a socket is set up<br> > for relaying the media. The RTP/UDPTL is just copied across the socket.<br> ><br> > Mode 3) No Media Mode. In this mode the SDPs are copied across and media is<br> > setup end to end... No media comes into freeswitch...<br> ><br> > All 3 Modes are more efficient and mode scalable then some other well known<br> > software. This is not necessarily due to the SIP stack in use, but more from<br> > a design perspective of taken from what Anthm and crew learned over the<br> > years as not to do.<br> ><br> ><br> ><br> ><br> >> From: Kurt Marasco <<a href="mailto:kmarasco@faithwork.org">kmarasco@faithwork.org</a>><br> >> Reply-To: <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br> >> Date: Sun, 16 Mar 2008 20:03:01 -0700<br> >> To: <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br> >> Subject: [Freeswitch-users] Conceptual Question about Freeswitch and SIP<br> >><br> >> It's my understanding that when I call from one SIP address to another<br> >> that Freeswitch manages the invites and then directly connects the two<br> >> sip devices, such that freeswitch is no longer involved in the<br> >> conversation. Is this how it is supposed to work?<br> >><br> >> The reason for my question is that the rtp traffic is going through my<br> >> freeswitch ip address at all times. It seems that both ends are speaking<br> >> g711u, so I don't believe that any transcoding is going on.<br> >><br> >> I must be missing something because having all data flow through<br> >> Freeswitch would not be scalable and I know that one of the big<br> >> differences that Freeswitch offers is scalability.<br> >><br> >> What am I missing....and I know that I am:)<br> >> Kurt<br> >><br> >> _______________________________________________<br> >> Freeswitch-users mailing list<br> >> <a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br> >> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br> >> UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br> >> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br> >><br> ><br> ><br> ><br> > _______________________________________________<br> > Freeswitch-users mailing list<br> > <a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br> > <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br> > UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br> > <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br> ><br> <br> _______________________________________________<br> Freeswitch-users mailing list<br> <a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br> UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br> </div></div></blockquote></div><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></blockquote></div><br></div></body></html>