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

[Stanislav Sinyagin]

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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