[Freeswitch-users] handling signalling externally, RTP via freeswitch

[Kees Jan Koster]

Dear Stanislav,

No, the audio will be bi-directional. Other than the custom signalling it will be a regular voice call.

Kees Jan


On 1 Jul 2014, at 14:31, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:

> Kees, do I understand it correctly that you're only sending audio towards the SIP endpoint, so it's not a bidirectional conversation?
> 
> In this case, you can utilize audio streaming via HTTP, probably it's the easiest way to deliver audio toward FreeSWITCH:
> 
> session.streamFile("shout://some.server.com/file.mp3", "")
> https://wiki.freeswitch.org/wiki/Mod_shout
> 
> 
> 
> 
> From: Kees Jan Koster <kjkoster at java-monitor.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>; ssinyagin at yahoo.com 
> Sent: Tuesday, July 1, 2014 1:27 PM
> Subject: Re: [Freeswitch-users] handling signalling externally, RTP via freeswitch
> 
> Dear Stanislav,
> 
> I was rather hoping to avoid that path. My C is not all that great. :-/ But if I have to...
> 
> Any endpoints that come close that I might want to look at?
> 
> Kees Jan
> 
> > AFAIK the RTP parameters and streaming are handled from the endpoint module:
> > https://wiki.freeswitch.org/wiki/Modules
> > 
> > so, you might end up in writing your own module for what you want.
> > 
> > 
> > From: Kees Jan Koster <kjkoster at kjkoster.org>
> > To: freeswitch-users at lists.freeswitch.org 
> > Sent: Tuesday, July 1, 2014 9:21 AM
> > Subject: [Freeswitch-users] handling signalling externally,    RTP via freeswitch
> > 
> > Dear All,
> > 
> > I need some reading advice/pointers on the following: I want to use FreeSwitch as a bridge from a custom client to SIP. The SIP part is easy and works. The non-SIP part is the challenge.
> > 
> > I have my audio sending client hooked up via mod_event_socket. I really like the way events and ESL work in FreeSwitch, by the way. Good control and easy to code. But I digress. So I have my client hooked up over ESL for signalling and need to know where to send the RTP stream to from the non-SIP client side.
> > 
> > Call flow (A-leg is the custom client, B-leg is sofia-SIP)
> >  [1] custom client starts call to SIP side over ESL [1] 
> >      FreeSwitch handles the signalling with the SIP client
> >      SIP client answers the call
> >  [2] FreeSwitch sends event which tells the client where the RTP stream should go to
> >      custom client starts sending RTP audio
> >      FreeSwitch bridges the two legs and people talk
> >      either side hangs up and the call is done
> > 
> > The reverse (SIP dialling the custom client) won't happen and does not have to be supported.
> > 
> > My concrete question is how [1] should happen. The custom client should issue an "bgapi originate". The arguments for the SIP side are clear, but what to use for the custom client? What should I fill in for [1], below?
> > 
> >  bgapi originate [1] sofia/internal/1002 at example.com
> > 
> > --
> > Kees Jan
> > 
> > http://java-monitor.com/
> > kjkoster at kjkoster.org
> > +31651838192
> > 
> > The secret of success lies in the stability of the goal. -- Benjamin Disraeli
> > 
> > 
> > _________________________________________________________________________
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> > 
> > 
> > 
> > 
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> > 
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> > 
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
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> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
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> 
> 
> --
> Kees Jan
> 
> kjkoster at java-monitor.com
> 
> http://java-monitor.com/
> +31651838192
> 
> The secret of success lies in the stability of the goal. -- Benjamin Disraeli
> 
> 


--
Kees Jan

http://java-monitor.com/
kjkoster at kjkoster.org
+31651838192

Change is good. Granted, it is good in retrospect, but change is good.