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>
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>><br>
><br>
><br>
><br>
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</div></div></blockquote></div><br>