26 Feb
2009
26 Feb
'09
2:19 p.m.
Thanks for that. So does it mean by using rtpproxy you will therefore carry all the rtp
streams through that server or can it be redirected to the sip provider from the
endpoint?
Also how do you put the kamailio server in the equation? Do you set it up as an external
proxy for the clients or do you register the clients to it then just use asterisk for the
media/vmail etc?
--- On Thu, 26/2/09, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote:
From: Daniel-Constantin Mierla
<miconda(a)gmail.com>
Subject: Re: [Kamailio-Users] Kamailio Newb questions
To: c_lougher(a)yahoo.co.uk
Cc: users(a)lists.kamailio.org
Date: Thursday, 26 February, 2009, 9:16 AM
Hello,
On 02/26/2009 12:59 AM, carl Lougher wrote:
yes it is possible. nathelper+rtpproxy is the option I use
and prefer
because of flexibility and performances. You can see an
example at:
http://www.voip-info.org/wiki/view/Kamailio+1.5.x+and+RTPProxy
through the rtpproxy server, which can be located on same
or different
machine than kamailio.
Yes, you can register to kamailio, see registrar and usrloc
modules.
Cheers,
Daniel
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
--
Daniel-Constantin Mierla
http://www.asipto.com
Howdy,
I'm trying to remove the media/rtp streams from an
asterisk server for natted
users so would like to know if
this is possible with kamailio.
Qu's:
What is the best option?
rtpproxy/mediaproxy?
nathelper?
If i use kamailio to achieve this does it mean that i
still have to carry the rtp
streams through the kamailio
server instead?
Also will i need to change the logon info for the
clients so they now logon to kamailio then i just point
registrar to asterisk?
Can i use kamailio for sip trunks to asterisk and
carry rtp and natted clients
media streams rather than
register to asterisk?
Many thanks,
Taff..
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