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> Reply-To: <freeswitch-users@lists.freeswitch.org> Date: Sun, 16 Mar 2008 20:03:01 -0700 To: <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
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