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

[Stanislav Sinyagin]

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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--
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
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