Re: [Kamailio-Users] how to know if the next gw is up? - sr-users

27 Feb 2009

Please direct your questions to users mailing lists. Private messages of 
such type are not honored very fast, they usually end in my very long 
unread mails pool.

See the dispatcher module, maybe it helps you.

Thanks,
Daniel

On 02/27/2009 11:18 AM, BERGANZ François wrote:
...
  Hello,

 I have 1SER and 5 Asterisk.
 I need to choose the dst with my algo...
 But, I need to have my server list and choose a server which is online.
 The best thing could be to have OPTIONS send to my asterisk and write responses in the
database!

 Have you an idea?

 Cordialement,
 BERGANZ François
  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

 -----Message d'origine-----
 De : users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] De la
part de Daniel-Constantin Mierla
 Envoyé : vendredi 27 février 2009 09:10
 À : c_lougher(a)yahoo.co.uk
 Cc : users(a)lists.kamailio.org
 Objet : Re: [Kamailio-Users] Kamailio Newb questions

 On 02/27/2009 12:07 AM, carl Lougher wrote:

  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?

       nathelper/rtpproxy does rtp relay on server, so the rtp goes: [caller] 
 === [rtpproxy] === [callee]

 In case of dealing with symmetric NAT, it is the only feasible way now 
 to make it work.

 If STUN is used, the rtp goes: [caller] === [callee] . But does not work 
 for symmetric nat.

 The best is to combine both of them so you get as less as possible RTP 
 relaying on server.

 Cheers,
 Daniel

  What are you using rtpproxy for that is different
than stun?

 --- On Thu, 26/2/09, Daniel-Constantin Mierla &lt;miconda(a)gmail.com&gt; wrote:

  From: Daniel-Constantin Mierla
&lt;miconda(a)gmail.com&gt;
 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

           &lt;miconda(a)gmail.com&gt; wrote:

          >> From: Daniel-Constantin Mierla
>>         
>>           
 &lt;miconda(a)gmail.com&gt;

 > 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

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

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           -- Daniel-Constantin Mierla
 http://www.asipto.com

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-- 
Daniel-Constantin Mierla
http://www.asipto.com