[Freeswitch-users] Conceptual Question about Freeswitch and SIP

[David Knell]

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 at faithwork.org 
> <mailto: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
>     <mailto:kmarasco at faithwork.org>>
>     >> Reply-To: <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>     >> Date: Sun, 16 Mar 2008 20:03:01 -0700
>     >> To: <freeswitch-users at lists.freeswitch.org
>     <mailto: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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-- 
David Knell, Director, 3C Limited
T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
http://www.3c.co.uk 

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