[Freeswitch-users] Conceptual Question about Freeswitch and SIP

[Michael Jerris]

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.

Mike


On Mar 17, 2008, at 8:51 AM, Vijay Ramnarine wrote:

> Would FS be able to do any sort of tracking / billing in No Media  
> Mode ?
> how about something like rtp timeout?
>
> On Sun, Mar 16, 2008 at 11:46 PM, Kurt Marasco  
> <kmarasco at 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 at faithwork.org>
> >> Reply-To: <freeswitch-users at lists.freeswitch.org>
> >> Date: Sun, 16 Mar 2008 20:03:01 -0700
> >> To: <freeswitch-users at 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
> >>
> >> _______________________________________________
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> >>
> >
> >
> >
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>
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