[Freeswitch-users] Free switch to ONLY handle RTP?
ssinyagin at gmail.com
Wed Oct 15 10:21:37 MSD 2014
isn't it easier to replace A+B?
as far as I understand, you would have to write your own module in C,
because the SDP and media manipulation functions are only exposed to the
internal C API.
So, it would be a custom development for just one installation, and after
you decide to leave, no-one will be able to support it. Why not just
replacing the whole telephony system at once?
On Mon, Oct 13, 2014 at 5:00 AM, Malcolm Lockyer <malcolm.lockyer at gmail.com>
> 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.
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> Official FreeSWITCH Sites
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
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