You can do RTP directly between the end points... You can do that 2 ways
- on your sofia profile, you’ll see ‘inbound-no-media’ set that to true and all calls on that profile will work in more like ser while still being a B2BUA
- the name of the channel variable escapes me at the moment but this can also be set in the dialplan prior to calling the bridge app so you can selectively bypass media
From: Krzysiek <cris7@o2.pl>
Reply-To: <freeswitch-users@lists.freeswitch.org>
Date: Sun, 20 Apr 2008 17:16:41 +0200
To: <freeswitch-users@lists.freeswitch.org>
Subject: Re: [Freeswitch-users] freeswitch does not listen on the SIP port?
Hi
I had to manually set my ip address in vars.xml and sip_profiles->default.xml and it now works well.
But I have another question... :)
I would like to set RTP channel directly between endpoints (like proxy server does). is it possible in Freeswitch? I guess, it is :).
And one more question...
Is Freeswitch the only one free/opensource solution that supports SRTP and TLS ?
I know that OpenSer and SipX have only TLS support (Sipx only on trunks?).
Thanks for help
Chris
----- Original Message -----
From: Brian West <mailto:brian@freeswitch.org>
To: freeswitch-users@lists.freeswitch.org
Sent: Friday, April 18, 2008 10:26 PM
Subject: Re: [Freeswitch-users] freeswitch does not listen on the SIP port?
It should just work unless you happened to have something else on 5060.
what is the output of "sofia status"
You might need to contact Andrew about this.
/b
On Apr 18, 2008, at 1:54 PM, Krzysiek wrote:
Hi there
I've installed trixswitch lately and i have a problem. I haven't changed anything in the configuration files in the conf directory yet. I see that freeswitch daemon is running. But I see that it isn't listening on the SIP Port 5060? Do I have to do something special to force the Freeswitch to listen on that port?
TIA for any help
Chris
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