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:
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(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:
>
>
>> 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..
>>
>>
>>
>>
>>
>>
>>
>>
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>>
>>
>>
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>
>
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>
>
-- Daniel-Constantin Mierla
http://www.asipto.com
_______________________________________________
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-- Daniel-Constantin Mierla
http://www.asipto.com
_______________________________________________
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