You can re-use some of mod_sofia's functions (like
sofia_glue_parse_sdp) and only write the part of signalling thats
different from SIP.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
[email protected]
On 4-Dec-09, at 8:52 PM, Michael Giagnocavo wrote:
Yes I was just thinking that it might be simpler to just fixup the
SDP and just write some custom script to talk control to the backend
conference system than to write a whole endpoint module. Especially
cause you can do the fixup and control in a high level language
(even if you use C#, you’re going to end up playing with pointers
except the syntax will be more verbose). Then again, I have a
natural aversion to C so maybe it’s just me ;)
-Michael
From: [email protected] [mailto:[email protected]
] On Behalf Of Phillip Jones
Sent: Friday, December 04, 2009 3:59 PM
To: [email protected]
Subject: Re: [Freeswitch-users] Bridging to a non SIP based system
Ah guys - that was exactly the nudge I was looking for - I will take
a look at the other endpoint modules like mod_skypiax etc. I will
also look at the SDP - I see where you are going there - I might not
even need the conference in that case.
Question is - could I write an endpoint is C# !!! :)
Thanks again - that's a great help.
On Fri, Dec 4, 2009 at 5:16 PM, Michael Giagnocavo
<[email protected]> wrote:
I think you will need to sort out the signaling first, as you’ll
have to tell the conference system to accept which RTP streams for
which conferences, as well as tell it to transmit to your callers, no?
After that, then I would imagine you just need to do SDP rewriting
when a call hits FreeSWITCH.
-Michael
From: [email protected] [mailto:[email protected]
] On Behalf Of Phillip Jones
Sent: Friday, December 04, 2009 2:29 PM
To: [email protected]
Subject: [Freeswitch-users] Bridging to a non SIP based system
Hi All,
Every so often you have to ask a question - where you know so little
- it's hard to even now where to start. This is one of the times. I
am not expecting an full answer here, just a gentle nudge in right
direction to get me started.
What I have is a propriety IP based conference system - who want to
add the ability to have inbound PSTN callers join their conferences.
All their signaling is propriety - no SIP - but I do have access to
that signaling schema so can do some translation. Enough to get the
IP / Port & CODEC of the RTP stream. They use speex rtp sessions
over TCP.
So from an architectural point of view I am thinking of having the
callers enter a FS conference and than bridge that conference to
their IP based conference room. That would do it.
The problem is that because I can not bridge using SIP (through a
Sofia gateway) to that IP based conference system I am kind of lost.
But it seems reasonable that I should be able to get my head round
this, because I know the IP / Port & CODEC of the RTP stream.
But perhaps I missing a key bit of knowledge/understanding here.
I would be grateful for any advise here.
Thanks a lot,
Phil
_______________________________________________
FreeSWITCH-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_______________________________________________
FreeSWITCH-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_______________________________________________
FreeSWITCH-users mailing list
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
