[Freeswitch-users] Free switch to ONLY handle RTP?

Malcolm Lockyer malcolm.lockyer at gmail.com
Mon Oct 13 07:00:27 MSD 2014


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.


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