So if i just want to hook off the rtp stream between endpoint and sip provider then am i better off to go for a stun server than nathelper/rtpproxy?
What are you using rtpproxy for that is different than stun?
--- On Thu, 26/2/09, Daniel-Constantin Mierla miconda@gmail.com wrote:
From: Daniel-Constantin Mierla miconda@gmail.com Subject: Re: [Kamailio-Users] Kamailio Newb questions To: c_lougher@yahoo.co.uk Cc: users@lists.kamailio.org Date: Thursday, 26 February, 2009, 12:27 PM On 02/26/2009 01:19 PM, carl Lougher wrote:
Thanks for that. So does it mean by using rtpproxy you
will therefore carry all the rtp streams through that server yes, that is the role of RTPProxy - to proxy the RTP streams, therefore those go via the server.
If you want end-to-end RTP stream, then look at STUN, if the phones are not behind symmetric nat, it can help.
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?
I do everything in kamailio but the media services which i do with asterisk (vmail, ivr, ...) - authentication, registration, call routing is done in kamailio.
Cheers, Daniel
--- On Thu, 26/2/09, Daniel-Constantin Mierla
miconda@gmail.com wrote:
From: Daniel-Constantin Mierla
Subject: Re: [Kamailio-Users] Kamailio Newb
questions
To: c_lougher@yahoo.co.uk Cc: users@lists.kamailio.org Date: Thursday, 26 February, 2009, 9:16 AM Hello,
On 02/26/2009 12:59 AM, carl Lougher wrote:
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.
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
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?
through the rtpproxy server, which can be located
on same
or different machine than kamailio.
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?
Yes, you can register to kamailio, see registrar
and usrloc
modules.
Cheers, Daniel
Many thanks, Taff..
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org
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-- Daniel-Constantin Mierla http://www.asipto.com
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
-- Daniel-Constantin Mierla http://www.asipto.com