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

Phillip Jones pjintheusa at gmail.com
Fri Dec 4 14:58:52 PST 2009


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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/133343b9/attachment-0002.html 


More information about the FreeSWITCH-users mailing list