Just thought I'd throw this out to the list just in case somebody can get my started somewhere. I'm hoping to get FreeSwitch to ONLY handle the RTP/audio/media of a call while handling the SIP myself/elsewhere.

I'm trying to pull apart and open up a really old in-house system that has 2 separate components for VoIP traffic.

Part A

- Handles SIP/presence/other stuff.

- Commands Part B.

Part B

- Establishes and passes RTP data/mixes/conferences/plays/records etc. based on a partial SDP (session description protocol).

I'm hoping to build a new server C as a replacement for B. I intend to use the command socket interface with FreeSwitch. Since I'm building a new server, I could "fake" SIP messages or something like that with FreeSwitch (i.e. have my new service C interact with FreeSwitch SIP). But I'd rather just command it to open / bridge / etc. RTP connections.

Can anybody point me to something that might get me started, other than the obvious command socket interface and basic docs? I'm going to be messing around with this anyway - just wonder if somebody can suggest a head start.

Thanks!

-Malcolm.