[Freeswitch-users] Bridging to a non SIP based system

[Mathieu Rene]

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
mrene at avgs.ca




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: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Phillip Jones
> Sent: Friday, December 04, 2009 3:59 PM
> To: freeswitch-users at lists.freeswitch.org
> 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  
> <mgg at giagnocavo.net> 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: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Phillip Jones
> Sent: Friday, December 04, 2009 2:29 PM
> To: freeswitch-users at lists.freeswitch.org
> 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
>
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