27 Feb
2009
27 Feb
'09
9:07 a.m.
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(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, 12:27 PM
On 02/26/2009 01:19 PM, carl Lougher wrote:
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
>> From: Daniel-Constantin
Mierla
<miconda(a)gmail.com>
-- Daniel-Constantin Mierla
http://www.asipto.com
_______________________________________________
Kamailio (OpenSER) - Users mailing list
Users(a)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
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?
--- On Thu, 26/2/09, Daniel-Constantin Mierla
<miconda(a)gmail.com> wrote:
>
> 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:
>
>> 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(a)lists.kamailio.org
>>
>>
>
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users