Hi Vijay,
Would FS be able to do any sort of tracking / billing in No Media Mode ? how about something like rtp timeout? Yes and no, respectively, although the same effect as the latter can be achieved using a SIP session timer - have a look at conf/sip_profiles/default.xml
--Dave
On Sun, Mar 16, 2008 at 11:46 PM, Kurt Marasco <kmarasco@faithwork.org <mailto:kmarasco@faithwork.org>> wrote:
Thanks for your comments Ken. They were very helpful. This group is great. I always seem to get multiple responses:)
Ken Rice wrote: > FreeSwitch has 3 modes of operation when it comes to sip calls. Please Note > FreeSwitch is a B2BUA and not a proxy although it has proxy like modes. > > Mode 1) Full Media Interaction this is what you are seeing. You can > completely interact with the media stream allowing for sniffing the media, > transcoding, or injecting the media for various purposes. > > Mode 2) Media Proxy only mode. In this mode FreeSwitch just acts as a dumb > RTP proxy for things like Topology Hiding and NAT busting. SIP SDPs are > copied across with updates to where to send the media and a socket is set up > for relaying the media. The RTP/UDPTL is just copied across the socket. > > Mode 3) No Media Mode. In this mode the SDPs are copied across and media is > setup end to end... No media comes into freeswitch... > > All 3 Modes are more efficient and mode scalable then some other well known > software. This is not necessarily due to the SIP stack in use, but more from > a design perspective of taken from what Anthm and crew learned over the > years as not to do. > > > > >> From: Kurt Marasco <kmarasco@faithwork.org <mailto:kmarasco@faithwork.org>> >> Reply-To: <freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> >> Date: Sun, 16 Mar 2008 20:03:01 -0700 >> To: <freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> >> Subject: [Freeswitch-users] Conceptual Question about Freeswitch and SIP >> >> It's my understanding that when I call from one SIP address to another >> that Freeswitch manages the invites and then directly connects the two >> sip devices, such that freeswitch is no longer involved in the >> conversation. Is this how it is supposed to work? >> >> The reason for my question is that the rtp traffic is going through my >> freeswitch ip address at all times. It seems that both ends are speaking >> g711u, so I don't believe that any transcoding is going on. >> >> I must be missing something because having all data flow through >> Freeswitch would not be scalable and I know that one of the big >> differences that Freeswitch offers is scalability. >> >> What am I missing....and I know that I am:) >> Kurt >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.org <mailto: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 >> > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org <mailto: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 >
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